devjournal

------------------------------------------------------------
2019/05/25 takeshi:
Development and test using node v10.15.3

------------------------------------------------------------
2019/06/08 takeshi:

We needed to use pjproject built from source to permit to debug it as pjw_call_respond gets stuck.


To be able to generate addon.node SO file we needed to build pjsip with -pPIC by doing:

cd pjproject-2.7.2
cat > user.mak <<EOF
export CFLAGS += -fPIC -g
export LDFLAGS +=
EOF

./configure
make dep && make clean && make



After preparing binding.gyp to use pjproject built from source, build of addon was successful:

takeshi@takeshi-desktop:node-sip$ node-gyp configure
gyp info it worked if it ends with ok
gyp info using node-gyp@3.8.0
gyp info using node@10.15.3 | linux | x64
gyp info spawn /usr/bin/python2
gyp info spawn args [ '/home/takeshi/.nvm/versions/node/v10.15.3/lib/node_modules/node-gyp/gyp/gyp_main.py',
gyp info spawn args   'binding.gyp',
gyp info spawn args   '-f',
gyp info spawn args   'make',
gyp info spawn args   '-I',
gyp info spawn args   '/home/takeshi/src/git/themonks/node-sip/build/config.gypi',
gyp info spawn args   '-I',
gyp info spawn args   '/home/takeshi/.nvm/versions/node/v10.15.3/lib/node_modules/node-gyp/addon.gypi',
gyp info spawn args   '-I',
gyp info spawn args   '/home/takeshi/.node-gyp/10.15.3/include/node/common.gypi',
gyp info spawn args   '-Dlibrary=shared_library',
gyp info spawn args   '-Dvisibility=default',
gyp info spawn args   '-Dnode_root_dir=/home/takeshi/.node-gyp/10.15.3',
gyp info spawn args   '-Dnode_gyp_dir=/home/takeshi/.nvm/versions/node/v10.15.3/lib/node_modules/node-gyp',
gyp info spawn args   '-Dnode_lib_file=/home/takeshi/.node-gyp/10.15.3/<(target_arch)/node.lib',
gyp info spawn args   '-Dmodule_root_dir=/home/takeshi/src/git/themonks/node-sip',
gyp info spawn args   '-Dnode_engine=v8',
gyp info spawn args   '--depth=.',
gyp info spawn args   '--no-parallel',
gyp info spawn args   '--generator-output',
gyp info spawn args   'build',
gyp info spawn args   '-Goutput_dir=.' ]
ngyp info ok 
takeshi@takeshi-desktop:node-sip$ node-gyp build
gyp info it worked if it ends with ok
gyp info using node-gyp@3.8.0
gyp info using node@10.15.3 | linux | x64
gyp info spawn make
gyp info spawn args [ 'BUILDTYPE=Release', '-C', 'build' ]
make: Entering directory '/home/takeshi/src/git/themonks/node-sip/build'
  SOLINK_MODULE(target) Release/obj.target/addon.node
  COPY Release/addon.node
make: Leaving directory '/home/takeshi/src/git/themonks/node-sip/build'
gyp info ok 


Howevever, when trying to use it we got:

takeshi@takeshi-desktop:node-sip$ node test.js 
internal/modules/cjs/loader.js:730
  return process.dlopen(module, path.toNamespacedPath(filename));
                 ^

Error: /home/takeshi/src/git/themonks/node-sip/build/Release/addon.node: undefined symbol: srtp_crypto_policy_set_aes_cm_256_hmac_sha1_80
    at Object.Module._extensions..node (internal/modules/cjs/loader.js:730:18)
    at Module.load (internal/modules/cjs/loader.js:600:32)
    at tryModuleLoad (internal/modules/cjs/loader.js:539:12)
    at Function.Module._load (internal/modules/cjs/loader.js:531:3)
    at Module.require (internal/modules/cjs/loader.js:637:17)
    at require (internal/modules/cjs/helpers.js:22:18)
    at Object.<anonymous> (/home/takeshi/src/git/themonks/node-sip/index.js:1:77)
    at Module._compile (internal/modules/cjs/loader.js:701:30)
    at Object.Module._extensions..js (internal/modules/cjs/loader.js:712:10)
    at Module.load (internal/modules/cjs/loader.js:600:32)


This was solved by checking the pjproject from apt dependencies:

takeshi@takeshi-desktop:node-sip$ pkg-config --libs libpjproject
-lpjsua2 -lstdc++ -lpjsua -lpjsip-ua -lpjsip-simple -lpjsip -lpjmedia-codec -lpjmedia -lpjmedia-videodev -lpjmedia-audiodev -lpjmedia -lpjnath -lpjlib-util -lsrtp2 -lpj -lopus -lssl -lcrypto -luuid -lm -lrt -lpthread -lasound -lSDL2 -lavdevice -lavformat -lavcodec -lswscale -lavutil -lv4l2 -lopencore-amrnb -lopencore-amrwb -lvo-amrwbenc

and adding them to the binding.gyp link_settings.libraries.


It also complained about a function from libspeex which was not mentioned in the the pkg-config output. So we addded it.


Then, it failed with:

takeshi@takeshi-desktop:node-sip$ node test.js 
pjw_init thread_id=1319614176

22:39:16.787         os_core_unix.c !pjlib 2.7.2 for POSIX initialized
node: symbol lookup error: /home/takeshi/src/git/themonks/node-sip/build/Release/addon.node: undefined symbol: pj_cis_buf_init


We found this in libpjlib-util:

takeshi@takeshi-desktop:node-sip$ nm pjproject-2.7.2/pjlib-util/lib/libpjlib-util-x86_64-unknown-linux-gnu.a  |grep pj_cis
                 U pj_cis_add_str
                 U pj_cis_buf_init
                 U pj_cis_init
0000000000000170 T pj_cis_add_alpha
00000000000001f0 T pj_cis_add_cis
00000000000001a0 T pj_cis_add_num
0000000000000100 T pj_cis_add_range
00000000000001b0 T pj_cis_add_str
0000000000000000 T pj_cis_buf_init
0000000000000230 T pj_cis_del_range
0000000000000270 T pj_cis_del_str
0000000000000090 T pj_cis_dup
0000000000000040 T pj_cis_init
00000000000002b0 T pj_cis_invert
takeshi@takeshi-desktop:node-sip$ nm pjproject-2.7.2/pjlib-util/lib/libpjlib-util-x86_64-unknown-linux-gnu.a  |grep pj_cis_buf_init
                 U pj_cis_buf_init
0000000000000000 T pj_cis_buf_init


We solved the above by putting -lpjlib-util-x86_64-unknown-linux-gnu as the last library in the list of pjprojects libs.

------------------------------------------------------------
2019/06/08 takeshi:

Problem of having pjw_call_respond stuck was narrowed down to:
  pjsip/src/pjsip-ua/sip_inv.c 
in function
  pjsip_inv_answer

The code stops at this line:
  pjsip_dlg_inc_lock(inv->dlg);

We tried to use latest pjproject 2.8 but the same problem happens.

UPDATE:
  this is happening because pjsip get stuck in call to pjsip_mutex_lock.
  It seems the cause is because we are using a separate thread (boost/thread) to do periodic poll of events
So we changed to make js to periodically do the poll and after that pjw_call_respond worked as expected.



Obs: while investigating this issue, we found this:

#####
Deadlocks
A deadlock is a situation wherein two or more competing actions are waiting for the other to finish, and thus neither ever does.

The internal PJSIP synchronization should be free from deadlock problem. However, deadlock may still arise when PJSIP has to interract with application's mutex.

To avoid deadlock introduced by application's mutex, the following rule MUST be obeyed at all times:

Mutexes MUST always be acquired with the same order, and the order is:
acquire PJSIP mutex first, then
acquire application mutex.
###

https://www.pjsip.org/locking.htm

However, to confirm this is not related to the problem we replaced this:

#define PJW_LOCK()      pthread_mutex_lock(&g_mutex)
#define PJW_UNLOCK()    pthread_mutex_unlock(&g_mutex)

with this:

#define PJW_LOCK()      printf("");
#define PJW_UNLOCK()    printf("");

and the problem persisted.

------------------------------------------------------------
2019/06/08 takeshi:
Due to:
  https://trac.pjsip.org/repos/changeset/5942
we reverted from pjproject-2.8 to pjproject-2.7.2


------------------------------------------------------------
2019/06/08 takeshi:

Failure to compile due change in definition of struct pjmedia_port_info:

takeshi@takeshi-desktop:node-sip$ node-gyp build
gyp info it worked if it ends with ok
gyp info using node-gyp@3.8.0
gyp info using node@10.15.3 | linux | x64
gyp info spawn make
gyp info spawn args [ 'BUILDTYPE=Release', '-C', 'build' ]
make: Entering directory '/home/takeshi/src/git/themonks/node-sip/build'
  CC(target) Release/obj.target/addon/src/pjmedia/src/pjmedia/dtmfdet.o
../src/pjmedia/src/pjmedia/dtmfdet.c: In function ‘pjmedia_dtmfdet_create’:
../src/pjmedia/src/pjmedia/dtmfdet.c:90:29: warning: assignment from incompatible pointer type [-Wincompatible-pointer-types]
     dtmfdet->base.put_frame = &dtmfdet_put_frame;
                             ^
../src/pjmedia/src/pjmedia/dtmfdet.c: In function ‘dtmfdet_put_frame’:
../src/pjmedia/src/pjmedia/dtmfdet.c:115:19: error: ‘pjmedia_port_info {aka struct pjmedia_port_info}’ has no member named ‘samples_per_frame’
   dport->base.info.samples_per_frame);
                   ^
addon.target.mk:133: recipe for target 'Release/obj.target/addon/src/pjmedia/src/pjmedia/dtmfdet.o' failed
make: *** [Release/obj.target/addon/src/pjmedia/src/pjmedia/dtmfdet.o] Error 1
make: Leaving directory '/home/takeshi/src/git/themonks/node-sip/build'
gyp ERR! build error 
gyp ERR! stack Error: `make` failed with exit code: 2
gyp ERR! stack     at ChildProcess.onExit (/home/takeshi/.nvm/versions/node/v10.15.3/lib/node_modules/node-gyp/lib/build.js:262:23)
gyp ERR! stack     at ChildProcess.emit (events.js:189:13)
gyp ERR! stack     at Process.ChildProcess._handle.onexit (internal/child_process.js:248:12)
gyp ERR! System Linux 4.15.0-43-generic
gyp ERR! command "/home/takeshi/.nvm/versions/node/v10.15.3/bin/node" "/home/takeshi/.nvm/versions/node/v10.15.3/bin/node-gyp" "build"
gyp ERR! cwd /home/takeshi/src/git/themonks/node-sip
gyp ERR! node -v v10.15.3
gyp ERR! node-gyp -v v3.8.0
gyp ERR! not ok 

UPDATE: actually dtmfdet.c is not necessary as it was just a plain pjmedia sample. Same for simpleua_dtmfdet.c.
So these were removed from binding.gyp target sources list. 

------------------------------------------------------------
2020/08/02 takeshi:

Installation using 'npm install' tested on Ubuntu 18.04.
------------------------------------------------------------
2020/08/08 takeshi:

When trying to build on Debian 10 I got:

$ npm install
npm WARN lifecycle sip-lab@1.0.0~install: cannot run in wd sip-lab@1.0.0 ./install.sh (wd=/usr/local/src/git/github_MayamaTakeshi/sip-lab)
audited 124 packages in 2.618s
found 0 vulnerabilities

But the process exit without error:
$ echo $?
0

which is bad because the addon was left unbuilt.

I had to use:
  npm install --unsafe-perm
otherwise, it doesn't run the install.sh script

Then build went further but now I got:

  CXX(target) Release/obj.target/addon/src/addon.o                                                                                                                                                          
../src/addon.cpp: In function 'Napi::Object init(Napi::Env, Napi::Object)':                                                                                                                                 
../src/addon.cpp:1361:60: error: no matching function for call to 'Napi::Function::New(Napi::Env&, <unresolved overloaded function type>)'                                                                  
   exports.Set("shutdown", Napi::Function::New(env, shutdown));                                                                                                                                             
                                                            ^                                                                                                                                               
In file included from /usr/local/src/git/github_MayamaTakeshi/sip-lab/node_modules/node-addon-api/napi.h:2051,                                                                                              
                 from ../src/addon.cpp:3:                                                                                                                                                                   
/usr/local/src/git/github_MayamaTakeshi/sip-lab/node_modules/node-addon-api/napi-inl.h:1678:17: note: candidate: 'template<class Callable> static Napi::Function Napi::Function::New(napi_env, Callable, con
st char*, void*)'                                                                                                                                                                                           
 inline Function Function::New(napi_env env,                                                                                                                                                                
                 ^~~~~~~~                                                                                                                                                                                   
/usr/local/src/git/github_MayamaTakeshi/sip-lab/node_modules/node-addon-api/napi-inl.h:1678:17: note:   template argument deduction/substitution failed:                                                    
../src/addon.cpp:1361:60: note:   couldn't deduce template parameter 'Callable'                                                                                                                             
   exports.Set("shutdown", Napi::Function::New(env, shutdown));                                                                                                                                             
                                                            ^         

Suspecting the name shutdown was conflicting with something else, I renamed the function as 
  shutdown_
and after that, build went further.

But then I got:

/usr/bin/ld: cannot find -lsrtp
collect2: error: ld returned 1 exit status
make: *** [addon.target.mk:226: Release/obj.target/addon.node] Error 1
make: Leaving directory '/usr/local/src/git/github_MayamaTakeshi/sip-lab/build'

This seems to be because in Debian 10, srtp library changed from libsrtp to libsrtp2.
But, actually, we should not use OS libs and instead we should the ThirdParty ones.
And in the binding.gyp we were actually using this unnecessary line:

           '-lsrtp',

But we already got this one:

           '-L ../pjproject/third_party/lib',


So, I removed '-lsrtp' and built was successful but then when running the sample I got:


$ node samples/simple.js 
internal/modules/cjs/loader.js:730
  return process.dlopen(module, path.toNamespacedPath(filename));
                 ^

Error: /usr/local/src/git/github_MayamaTakeshi/sip-lab/build/Release/addon.node: undefined symbol: crypto_policy_set_aes_cm_256_hmac_sha1_80
    at Object.Module._extensions..node (internal/modules/cjs/loader.js:730:18)
    at Module.load (internal/modules/cjs/loader.js:600:32)
    at tryModuleLoad (internal/modules/cjs/loader.js:539:12)
    at Function.Module._load (internal/modules/cjs/loader.js:531:3)
    at Module.require (internal/modules/cjs/loader.js:637:17)
    at require (internal/modules/cjs/helpers.js:22:18)
    at Object.<anonymous> (/usr/local/src/git/github_MayamaTakeshi/sip-lab/index.js:1:77)
    at Module._compile (internal/modules/cjs/loader.js:701:30)
    at Object.Module._extensions..js (internal/modules/cjs/loader.js:712:10)
    at Module.load (internal/modules/cjs/loader.js:600:32)


So I disable SRTP by setting:

#define PJMEDIA_HAS_SRTP  0

in config_site.h


However, when trying to run
  node samples/simple.js
we are frequently (almost all the time) getting this:

$ node samples/simple.js 
pjw_init thread_id=583253000

21:12:20.187 os_core_unix.c !pjlib 2.5.5-svn for POSIX initialized
undefined
t1 { id: 0, ip: '127.0.0.1', port: 5090 }
t2 { id: 1, ip: '127.0.0.1', port: 5092 }
inv=22c3c118 tdata=3779318

on_state_changed

status=583254296

21:12:20.260                !: Success
wait (line /usr/local/src/git/github_MayamaTakeshi/sip-lab/samples/simple.js:24) started. Waiting for expected_events:
[
  partial_match({
    event: 'incoming_call',
    call_id: collect['call_id']()
  }),
  partial_match({
    event: 'response',
    call_id: 0,
    method: 'INVITE',
    msg: sip_msg({
      $rs: '100',
      $rr: 'Trying',
      $(hdrcnt(via)): 1,
      $hdr(call-id): collect['sip_call_id'](),
      $fU: 'a',
      $fd: 't',
      $tU: 'b',
      $hdr(l): '0'
    })
  })
]
Segmentation fault



Checking with gdb:
Thread 1 "node" received signal SIGSEGV, Segmentation fault.
__strlen_sse2 () at ../sysdeps/x86_64/multiarch/../strlen.S:120
120     ../sysdeps/x86_64/multiarch/../strlen.S: No such file or directory.
(gdb) bt                     
#0  0x00007ffff7b40206 in __strlen_sse2 () at ../sysdeps/x86_64/multiarch/../strlen.S:120
#1  0x00007ffff7af99ef in _IO_vfprintf_internal (s=0x7fffffff6af0, format=0x7ffff52533ec "on_rx_request %.*s\n\n", ap=0x7fffffff91b0) at vfprintf.c:1638
#2  0x00007ffff7afa866 in buffered_vfprintf (s=s@entry=0x7ffff7c64760 <_IO_2_1_stdout_>, format=format@entry=0x7ffff52533ec "on_rx_request %.*s\n\n", args=args@entry=0x7fffffff91b0) at vfprintf.c:2322
#3  0x00007ffff7af7eb2 in _IO_vfprintf_internal (s=0x7ffff7c64760 <_IO_2_1_stdout_>, format=0x7ffff52533ec "on_rx_request %.*s\n\n", ap=ap@entry=0x7fffffff91b0) at vfprintf.c:1296
#4  0x00007ffff7b00606 in __printf (format=<optimized out>) at printf.c:33
#5  0x00007ffff51ccbe6 in _addon_log(int, char const*, ...) () at /usr/local/src/git/github_MayamaTakeshi/sip-lab/build/Release/addon.node
#6  0x00007ffff51d81a9 in on_rx_request(pjsip_rx_data*) () at /usr/local/src/git/github_MayamaTakeshi/sip-lab/build/Release/addon.node
#7  0x00007ffff520b94b in pjsip_endpt_process_rx_data (p_handled=0x7fffffffa75c, p=0x7fffffffa780, rdata=0x266f488, endpt=0x261e7a8) at ../src/pjsip/sip_endpoint.c:887
#8  0x00007ffff520b94b in pjsip_endpt_process_rx_data (endpt=0x261e7a8, rdata=0x266f488, p=<optimized out>, p_handled=0x7fffffffa75c) at ../src/pjsip/sip_endpoint.c:824
#9  0x00007ffff520bb56 in endpt_on_rx_msg (endpt=0x261e7a8, status=<optimized out>, rdata=0x266f488) at ../src/pjsip/sip_endpoint.c:1037
#10 0x00007ffff52117a6 in pjsip_tpmgr_receive_packet (mgr=<optimized out>, rdata=rdata@entry=0x266f488) at ../src/pjsip/sip_transport.c:1938
#11 0x00007ffff5213a4f in udp_on_read_complete (key=0x26257e8, op_key=<optimized out>, bytes_read=<optimized out>) at ../src/pjsip/sip_transport_udp.c:170
#12 0x00007ffff5236e28 in ioqueue_dispatch_read_event (h=0x26257e8, ioqueue=0x2629600) at ../src/pj/ioqueue_common_abs.c:605
#13 0x00007ffff5236e28 in ioqueue_dispatch_read_event (ioqueue=0x2629600, h=0x26257e8) at ../src/pj/ioqueue_common_abs.c:433
#14 0x00007ffff52386cf in pj_ioqueue_poll (ioqueue=0x2629600, timeout=timeout@entry=0x7fffffffb190) at ../src/pj/ioqueue_select.c:981
#15 0x00007ffff520b70a in pjsip_endpt_handle_events2 (endpt=0x261e7a8, max_timeout=0x7fffffffb1d0, p_count=0x0) at ../src/pjsip/sip_endpoint.c:742
#16 0x00007ffff51cf824 in handle_events() () at /usr/local/src/git/github_MayamaTakeshi/sip-lab/build/Release/addon.node
#17 0x00007ffff51d00a4 in __pjw_poll(char*) () at /usr/local/src/git/github_MayamaTakeshi/sip-lab/build/Release/addon.node
#18 0x00007ffff51dab71 in do_poll(Napi::CallbackInfo const&) () at /usr/local/src/git/github_MayamaTakeshi/sip-lab/build/Release/addon.node
#19 0x00007ffff51e4923 in Napi::details::CallbackData<Napi::Value (*)(Napi::CallbackInfo const&), Napi::Value>::Wrapper(napi_env__*, napi_callback_info__*) ()
    at /usr/local/src/git/github_MayamaTakeshi/sip-lab/build/Release/addon.node
#20 0x00000000008e85b5 in (anonymous namespace)::v8impl::FunctionCallbackWrapper::Invoke(v8::FunctionCallbackInfo<v8::Value> const&) ()
#21 0x0000000000b62a3f in v8::internal::MaybeHandle<v8::internal::Object> v8::internal::(anonymous namespace)::HandleApiCallHelper<false>(v8::internal::Isolate*, v8::internal::Handle<v8::internal::HeapObj
ect>, v8::internal::Handle<v8::internal::HeapObject>, v8::internal::Handle<v8::internal::FunctionTemplateInfo>, v8::internal::Handle<v8::internal::Object>, v8::internal::BuiltinArguments) ()
#22 0x0000000000b635a9 in v8::internal::Builtin_HandleApiCall(int, v8::internal::Object**, v8::internal::Isolate*) ()
#23 0x0000309d62fdbe1d in  ()


So it is a bug in our code.


But building it with these changes in on ubuntu 18.04.4 doesn't cause this crash which means in Debian 10 a condition that increases the chances of crash was introduced.
Anyway, we will commit these changes and solve the crash later.

------------------------------------------------------------
2020/08/09 takeshi:

We had to disable SRTP pjsip support (config_site.h) because of this:

$ node samples/simple.js 
internal/modules/cjs/loader.js:730
  return process.dlopen(module, path.toNamespacedPath(filename));
                 ^

Error: /mnt/backup/takeshi_tmp/sip-lab/build/Release/addon.node: undefined symbol: crypto_policy_set_aes_cm_256_hmac_sha1_80
    at Object.Module._extensions..node (internal/modules/cjs/loader.js:730:18)
    at Module.load (internal/modules/cjs/loader.js:600:32)
    at tryModuleLoad (internal/modules/cjs/loader.js:539:12)
    at Function.Module._load (internal/modules/cjs/loader.js:531:3)
    at Module.require (internal/modules/cjs/loader.js:637:17)
    at require (internal/modules/cjs/helpers.js:22:18)
    at Object.<anonymous> (/mnt/backup/takeshi_tmp/sip-lab/index.js:1:77)
    at Module._compile (internal/modules/cjs/loader.js:701:30)
    at Object.Module._extensions..js (internal/modules/cjs/loader.js:712:10)
    at Module.load (internal/modules/cjs/loader.js:600:32)

------------------------------------------------------------
2020/08/12 takeshi:

We did adjustments to support late negotiation.
For this we split original callback on_rx_offer in new callbacks on_rx_offer and on_rx_reinvite.
This worked with initial INVITE late negotiation.
However, it crashed with RE-INVITE late negotiation.
The problem is that on_rx_reinvite is getting an offer without media (due RE-INVITE without SDP).
Instead, I think we should go back to use original on_rx_offer because it seems this is called only when we get media offer and so we would avoid this problem.


------------------------------------------------------------
2022/08/12 takeshi:

When rebuilding the addon, this should be enough (it should be fast):
```
  npm install --unsafe-perm
```
------------------------------------------------------------
2022/09/11 takeshi:

To try to solve #21, we upraded pjproject to commit 797088ed133c98492519b7d042b75735f6f9388c.
However, after doing it we verified sending CANCEL doesn't cause the other side to send '497 Request Terminated' by itself anymore (see samples/sip_cancel.js)
 

------------------------------------------------------------
2022/09/17 takeshi:
We will start shipping prebuilt binaries of the addon.
We create them by doing:
```
prebuildify --strip -t 15.0.0 -t 16.0.0 -t 17.0.0 -t 18.0.0
```

2022/09/18 takeshi:

Tried to statically link to openssl but then prebuilt file increased from:
```
$ ls -l prebuilds/linux-x64/        
total 7240
-rwxr-xr-x 1 root root 1858744 Sep 17 14:04 node.abi102.node
-rwxr-xr-x 1 root root 1858744 Sep 17 14:04 node.abi108.node
-rwxr-xr-x 1 root root 1846424 Sep 17 14:04 node.abi88.node
-rwxr-xr-x 1 root root 1846424 Sep 17 14:04 node.abi93.node
```
to
```
$ ls -l prebuilds/linux-x64/
total 27328
-rwxr-xr-x 1 root root 6994712 Sep 18 08:58 node.abi102.node
-rwxr-xr-x 1 root root 6994712 Sep 18 08:58 node.abi108.node
-rwxr-xr-x 1 root root 6994680 Sep 18 08:58 node.abi88.node
-rwxr-xr-x 1 root root 6994680 Sep 18 08:58 node.abi93.node
```

For now, I will keep linking openssl so libs.

------------------------------------------------------------
2022/09/17 takeshi:

For reference, the addon doesn't work on Debian 8:

```
$ node samples/delayed_media.js 
node:internal/modules/cjs/loader:1210
  return process.dlopen(module, path.toNamespacedPath(filename));
                 ^

Error: /lib/x86_64-linux-gnu/libm.so.6: version `GLIBC_2.27' not found (required by /root/tmp/sip-lab/prebuilds/linux-x64/node.abi93.node)
    at Object.Module._extensions..node (node:internal/modules/cjs/loader:1210:18)
    at Module.load (node:internal/modules/cjs/loader:1004:32)
    at Function.Module._load (node:internal/modules/cjs/loader:839:12)
    at Module.require (node:internal/modules/cjs/loader:1028:19)
    at require (node:internal/modules/cjs/helpers:102:18)
    at load (/root/tmp/sip-lab/node_modules/node-gyp-build/index.js:22:10)
    at Object.<anonymous> (/root/tmp/sip-lab/index.js:1:40)
    at Module._compile (node:internal/modules/cjs/loader:1126:14)
    at Object.Module._extensions..js (node:internal/modules/cjs/loader:1180:10)
    at Module.load (node:internal/modules/cjs/loader:1004:32) {
  code: 'ERR_DLOPEN_FAILED'
}
```

------------------------------------------------------------
2022/09/17 takeshi:

To force a build of the addon when a prebuilt is not available, we need to add an 'actions' list for the targe in the binding.gyp file.
We followed a tip from here:
  https://stackoverflow.com/questions/27275340/run-configure-in-gyp-file

------------------------------------------------------------
2022/11/12 takeshi:

Building docker images to use prebuildify-cross:

Smoke test:

cd docker-images/debian10
cp ../../build_deps.sh .
docker build -t mayamatakeshi/sip-lab-debian10:latest .
cd ../..
npx prebuildify-cross -i mayamatakeshi/sip-lab-debian10:latest -t 16.0.0 --strip

The above worked and the resulting addon:

takeshi@takeshi-desktop:~/src/git/MayamaTakeshi/sip-lab$ file prebuilds/linux-x64/node.abi93.node 
prebuilds/linux-x64/node.abi93.node: ELF 64-bit LSB shared object, x86-64, version 1 (GNU/Linux), dynamically linked, BuildID[sha1]=16ca20da2663c49f1f232e7d99b46709d4f57285, stripped

could be used in both debian10 and debian11.

Then I tried with Ubuntu 22.04:

cd docker-images/jammy
cp ../../build_deps.sh .
docker build -t mayamatakeshi/sip-lab-jammy:latest .
cd ../..
npx prebuildify-cross -i mayamatakeshi/sip-lab-jammy:latest -t 16.0.0 --strip

and it generated an addon with the same path/name:

takeshi@takeshi-desktop:~/src/git/MayamaTakeshi/sip-lab$ file prebuilds/linux-x64/node.abi93.node 
prebuilds/linux-x64/node.abi93.node: ELF 64-bit LSB shared object, x86-64, version 1 (GNU/Linux), dynamically linked, BuildID[sha1]=6587c79403de026748fccf7ec224d75426d20892, stripped

But when I tried to use it on debian11 I got:

[root@lab225201-basix sip-lab]$ node samples/simple.js 
node:internal/modules/cjs/loader:1183
  return process.dlopen(module, path.toNamespacedPath(filename));
                 ^

Error: libssl.so.3: cannot open shared object file: No such file or directory
    at Object.Module._extensions..node (node:internal/modules/cjs/loader:1183:18)
    at Module.load (node:internal/modules/cjs/loader:981:32)
    at Function.Module._load (node:internal/modules/cjs/loader:822:12)
    at Module.require (node:internal/modules/cjs/loader:1005:19)
    at require (node:internal/modules/cjs/helpers:102:18)
    at load (/root/tmp/sip-lab/node_modules/node-gyp-build/index.js:22:10)
    at Object.<anonymous> (/root/tmp/sip-lab/index.js:1:40)
    at Module._compile (node:internal/modules/cjs/loader:1101:14)
    at Object.Module._extensions..js (node:internal/modules/cjs/loader:1153:10)
    at Module.load (node:internal/modules/cjs/loader:981:32) {
  code: 'ERR_DLOPEN_FAILED'
}

So, the problem is that there is no way to differentiate between distros and distro versions. We can only identify the platform like linux-x64.

What we can try to do is to build all dependencies and link to them statically so to not depend on what is installed in the OS:

takeshi@takeshi-desktop:~/src/git/MayamaTakeshi/sip-lab$ ldd prebuilds/linux-x64/node.abi93.node 
	linux-vdso.so.1 (0x00007fff6f751000)
	libstdc++.so.6 => /lib/x86_64-linux-gnu/libstdc++.so.6 (0x00007f0749f35000)
	libssl.so.3 => /lib/x86_64-linux-gnu/libssl.so.3 (0x00007f0749e91000)
	libcrypto.so.3 => /lib/x86_64-linux-gnu/libcrypto.so.3 (0x00007f0749a4f000)
	libuuid.so.1 => /lib/x86_64-linux-gnu/libuuid.so.1 (0x00007f0749a46000)
	libtiff.so.5 => /lib/x86_64-linux-gnu/libtiff.so.5 (0x00007f07499c0000)
	libspeex.so.1 => /lib/x86_64-linux-gnu/libspeex.so.1 (0x00007f07499a0000)
	libm.so.6 => /lib/x86_64-linux-gnu/libm.so.6 (0x00007f07498b9000)
	libgcc_s.so.1 => /lib/x86_64-linux-gnu/libgcc_s.so.1 (0x00007f0749899000)
	libc.so.6 => /lib/x86_64-linux-gnu/libc.so.6 (0x00007f0749671000)
	/lib64/ld-linux-x86-64.so.2 (0x00007f074a34c000)
	libwebp.so.7 => /lib/x86_64-linux-gnu/libwebp.so.7 (0x00007f0749605000)
	libzstd.so.1 => /lib/x86_64-linux-gnu/libzstd.so.1 (0x00007f0749536000)
	liblzma.so.5 => /lib/x86_64-linux-gnu/liblzma.so.5 (0x00007f0749509000)
	libjbig.so.0 => /lib/x86_64-linux-gnu/libjbig.so.0 (0x00007f07494f8000)
	libjpeg.so.8 => /lib/x86_64-linux-gnu/libjpeg.so.8 (0x00007f0749477000)
	libdeflate.so.0 => /lib/x86_64-linux-gnu/libdeflate.so.0 (0x00007f0749453000)
	libz.so.1 => /lib/x86_64-linux-gnu/libz.so.1 (0x00007f0749437000)
takeshi@takeshi-desktop:~/src/git/MayamaTakeshi/sip-lab$ 

I think this will minimize problems but it will not completely eliminate them.

------------------------------------------------------------
2022/11/12 takeshi:

Here is what ldd on Debian11 outputs to an addon built on ubuntu 22.04:

[root@lab225201-basix tmp]$ ldd addon.node 
./addon.node: /lib/x86_64-linux-gnu/libstdc++.so.6: version `GLIBCXX_3.4.29' not found (required by ./addon.node)
./addon.node: /lib/x86_64-linux-gnu/libc.so.6: version `GLIBC_2.34' not found (required by ./addon.node)
./addon.node: /lib/x86_64-linux-gnu/libc.so.6: version `GLIBC_2.33' not found (required by ./addon.node)
	linux-vdso.so.1 (0x00007ffe98ecf000)
	libstdc++.so.6 => /lib/x86_64-linux-gnu/libstdc++.so.6 (0x00007fb8e8e57000)
	libssl.so.3 => not found
	libcrypto.so.3 => not found
	libuuid.so.1 => /lib/x86_64-linux-gnu/libuuid.so.1 (0x00007fb8e8e4e000)
	libtiff.so.5 => /lib/x86_64-linux-gnu/libtiff.so.5 (0x00007fb8e8dc8000)
	libspeex.so.1 => /lib/x86_64-linux-gnu/libspeex.so.1 (0x00007fb8e8dad000)
	libm.so.6 => /lib/x86_64-linux-gnu/libm.so.6 (0x00007fb8e8c67000)
	libgcc_s.so.1 => /lib/x86_64-linux-gnu/libgcc_s.so.1 (0x00007fb8e8c4d000)
	libc.so.6 => /lib/x86_64-linux-gnu/libc.so.6 (0x00007fb8e8a88000)
	/lib64/ld-linux-x86-64.so.2 (0x00007fb8e9208000)
	libwebp.so.6 => /lib/x86_64-linux-gnu/libwebp.so.6 (0x00007fb8e8a1f000)
	libzstd.so.1 => /lib/x86_64-linux-gnu/libzstd.so.1 (0x00007fb8e8944000)
	liblzma.so.5 => /lib/x86_64-linux-gnu/liblzma.so.5 (0x00007fb8e891c000)
	libjbig.so.0 => /lib/x86_64-linux-gnu/libjbig.so.0 (0x00007fb8e870c000)
	libjpeg.so.62 => /lib/x86_64-linux-gnu/libjpeg.so.62 (0x00007fb8e8688000)
	libdeflate.so.0 => /lib/x86_64-linux-gnu/libdeflate.so.0 (0x00007fb8e866c000)
	libz.so.1 => /lib/x86_64-linux-gnu/libz.so.1 (0x00007fb8e864f000)
	libpthread.so.0 => /lib/x86_64-linux-gnu/libpthread.so.0 (0x00007fb8e862d000)
[root@lab225201-basix tmp]$ 

So, if we build and statically link to libssl and libcrypto, we should avoid problems.


And here it is for old Centos 7:

[root@IPX029 tmp]# cat /etc/redhat-release 
CentOS Linux release 7.9.2009 (Core)
[root@IPX029 tmp]# ldd addon.node 
./addon.node: /usr/lib64/libm.so.6: version `GLIBC_2.27' not found (required by ./addon.node)
./addon.node: /usr/lib64/libstdc++.so.6: version `GLIBCXX_3.4.26' not found (required by ./addon.node)
./addon.node: /usr/lib64/libstdc++.so.6: version `CXXABI_1.3.8' not found (required by ./addon.node)
./addon.node: /usr/lib64/libstdc++.so.6: version `GLIBCXX_3.4.29' not found (required by ./addon.node)
./addon.node: /usr/lib64/libstdc++.so.6: version `GLIBCXX_3.4.21' not found (required by ./addon.node)
./addon.node: /usr/lib64/libstdc++.so.6: version `CXXABI_1.3.9' not found (required by ./addon.node)
./addon.node: /usr/lib64/libc.so.6: version `GLIBC_2.22' not found (required by ./addon.node)
./addon.node: /usr/lib64/libc.so.6: version `GLIBC_2.34' not found (required by ./addon.node)
./addon.node: /usr/lib64/libc.so.6: version `GLIBC_2.33' not found (required by ./addon.node)
	linux-vdso.so.1 =>  (0x00007fffa6cdf000)
	libstdc++.so.6 => /usr/lib64/libstdc++.so.6 (0x00007f8fd946e000)
	libssl.so.3 => not found
	libcrypto.so.3 => not found
	libuuid.so.1 => /usr/lib64/libuuid.so.1 (0x00007f8fd9269000)
	libtiff.so.5 => /usr/lib64/libtiff.so.5 (0x00007f8fd8ff5000)
	libspeex.so.1 => /usr/lib64/libspeex.so.1 (0x00007f8fd8ddc000)
	libm.so.6 => /usr/lib64/libm.so.6 (0x00007f8fd8ada000)
	libgcc_s.so.1 => /usr/lib64/libgcc_s.so.1 (0x00007f8fd88c4000)
	libc.so.6 => /usr/lib64/libc.so.6 (0x00007f8fd84f6000)
	/lib64/ld-linux-x86-64.so.2 (0x00007f8fd9776000)
	libjbig.so.2.0 => /usr/lib64/libjbig.so.2.0 (0x00007f8fd82ea000)
	libjpeg.so.62 => /usr/lib64/libjpeg.so.62 (0x00007f8fd8095000)
	libz.so.1 => /usr/lib64/libz.so.1 (0x00007f8fd7e7f000)
[root@IPX029 tmp]# 

Update: it will not be done. See #37.

------------------------------------------------------------
2023/01/05 takeshi:

I will need to add support for MRCP, MSRP etc.
For this, I should use pjlib networking functions so that they can be integrated with existing code using the same ioqueue in place for sip/rtp.
Sockets should be allocated in sip session pool so that they can be reclained when the session ends.

I asked ChatGPT:

"write a sample echo server using pjlib and ioqueue."

and it replied with:

"Here is a simple example of how you might implement an echo server using PJLIB and the ioqueue library:"

```
#include <pjlib.h>
#include <pjlib-util.h>
#include <pjnath.h>

#define PORT 4000

/* Callback called by the library when incoming data is received */
static pj_bool_t on_data_received(pj_activesock_t *asock,
                                   void *data,
                                   pj_size_t size,
                                   pj_status_t status,
                                   pj_size_t *remainder)
{
  /* Send the data back to the client */
  pj_ssize_t sent = pj_activesock_send(asock, data, size, 0);
  if (sent < 0) {
    /* Error sending data */
    return PJ_FALSE;
  } else {
    /* Successfully sent data, update the remainder */
    *remainder = size - sent;
    return PJ_TRUE;
  }
}

/* Callback called by the library when a new client connects */
static pj_bool_t on_accept_complete(pj_activesock_t *asock,
                                     pj_sock_t sock,
                                     const pj_sockaddr_t *src_addr,
                                     int src_addr_len)
{
  /* Start reading data from the client */
  pj_activesock_start_read(asock, NULL, 0, 0);
  return PJ_TRUE;
}

int main()
{
  pj_status_t status;
  pj_caching_pool cp;
  pj_pool_t *pool;
  pj_sock_t sock;
  pj_activesock_t *asock;

  /* Initialize PJLIB */
  status = pj_init();
  if (status != PJ_SUCCESS) {
    /* Error initializing PJLIB */
    return 1;
  }

  /* Create a pool factory */
  pj_caching_pool_init(&cp, NULL, 0);
  pool = pj_pool_create(&cp.factory, "echo-server", 1000, 1000, NULL);

  /* Create a socket */
  status = pj_sock_socket(PJ_AF_INET, PJ_SOCK_STREAM, 0, &sock);
  if (status != PJ_SUCCESS) {
    /* Error creating socket */
    return 1;
  }

  /* Bind the socket to a port */
  pj_sockaddr_in addr;
  pj_bzero(&addr, sizeof(addr));
  addr.sin_family = PJ_AF_INET;
  addr.sin_addr.s_addr = pj_htonl(PJ_INADDR_ANY);
  addr.sin_port = pj_htons(PORT);
  status = pj_sock_bind(sock, &addr, sizeof(addr));
  if (status != PJ_SUCCESS) {
    /* Error binding socket */
    return 1;
  }

  /* Start listening for incoming connections
  status = pj_sock_listen(sock, 5);
  if (status != PJ_SUCCESS) {
    /* Error listening for connections */
    return 1;
  }

  /* Create an active socket */
  pj_activesock_cfg asock_cfg;
  pj_activesock_cfg_default(&asock_cfg);
  asock_cfg.grp_lock = NULL;
  status = pj_activesock_create(pool, sock, pj_SOCK_STREAM(), &asock_cfg, &asock);
  if (status != PJ_SUCCESS) {
    /* Error creating active socket */
    return 1;
  }

  /* Set the callbacks for the active socket */
  pj_activesock_set_recv_cb(asock, pool, &on_data_received);
  pj_activesock_set_accept_cb(asock, pool, &on_accept_complete);

  /* Start accepting connections */
  pj_sockaddr_in local_addr;
  int addr_len = sizeof(local_addr);
  pj_sock_getsockname(sock, &local_addr, &addr_len);
  printf("Echo server listening on %s:%d\n",
         pj_inet_ntoa(local_addr.sin_addr),
         pj_ntohs(local_addr.sin_port));
  status = pj_activesock_start_accept(asock, pool);
  if (status != PJ_SUCCESS) {
    /* Error starting accept */
    return 1;
  }

  /* Run the event loop */
  pj_bool_t running= PJ_TRUE;
while (running) {
pj_time_val timeout = { 0, 10 };
pj_ioqueue_poll(pj_get_ioqueue_default(), &timeout);
}

/* Clean up */
pj_activesock_close(asock);
pj_sock_close(sock);
pj_pool_release(pool);
pj_caching_pool_destroy(&cp);
pj_shutdown();

return 0;
}

```

"This code creates a socket, binds it to a port, and starts listening for incoming connections. When a new connection is accepted, it creates an active socket and sets the callbacks for receiving data and accepting new connections. The event loop runs continuously, calling the ioqueue's poll function to process incoming data and events. When the server is no longer needed, it closes the active socket and the underlying socket, releases the memory pool, and shuts down PJLIB."


I eventually implemented working smoke test code by reading docs and samples and it seems the above indeed would work as it is very similar to what I came up with.
So although ChatGPT frequently generates bogus code with an authoritative tone, for some cases it seems to be usable.

------------------------------------------------------------
2022/01/06 takeshi:

Regarding on_rx_reinvite:

If we enable it like this:

    inv_cb.on_rx_offer = NULL;
    inv_cb.on_rx_offer2 = &on_rx_offer2;
    inv_cb.on_rx_reinvite = &on_rx_reinvite;

static pj_status_t on_rx_reinvite(pjsip_inv_session *inv, const pjmedia_sdp_session *offer, pjsip_rx_data *rdata) {
    printf("on_rx_reinvite\n");
    return PJ_SUCCESS;
}


then when we do a reinvite on a call like this:

sip.call.reinvite(oc.id, {hold: true})

we will get '200 OK' without a body like this:

INVITE sip:sip-lab@127.0.0.1:5092 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5090;rport;branch=z9hG4bKPj95c7dd9c-5f1f-489c-90b6-dd90d03bc905
Max-Forwards: 70
From: sip:alice@test.com;tag=86bee9d5-ca74-41a8-b135-8a7d6eda2de0
To: sip:bob@127.0.0.1;tag=b7808a74-9613-458a-82a1-20d033347001
Contact: <sip:sip@127.0.0.1:5090>
Call-ID: fd1e8d7f-1627-4fae-b886-c64b88670eaa
CSeq: 22618 INVITE
Allow: MESSAGE, SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE
Supported: replaces
Content-Type: application/sdp
Content-Length:   586

v=0
o=- 3881951753 3881951754 IN IP4 127.0.0.1
s=pjmedia
t=0 0
m=audio 10000 RTP/AVP 0 8 3 96 97 98 99 9 18 120 121 122
c=IN IP4 127.0.0.1
b=TIAS:64000
a=rtcp:10001 IN IP4 127.0.0.1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:96 iLBC/8000
a=fmtp:96 mode=30
a=rtpmap:97 speex/32000
a=rtpmap:98 speex/16000
a=rtpmap:99 speex/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=rtpmap:120 telephone-event/8000
a=fmtp:120 0-16
a=rtpmap:121 telephone-event/32000
a=fmtp:121 0-16
a=rtpmap:122 telephone-event/16000
a=fmtp:122 0-16
a=sendonly


SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.0.1:5090;rport=5090;received=127.0.0.1;branch=z9hG4bKPj95c7dd9c-5f1f-489c-90b6-dd90d03bc905
Call-ID: fd1e8d7f-1627-4fae-b886-c64b88670eaa
From: <sip:alice@test.com>;tag=86bee9d5-ca74-41a8-b135-8a7d6eda2de0
To: <sip:bob@127.0.0.1>;tag=b7808a74-9613-458a-82a1-20d033347001
CSeq: 22618 INVITE
Contact: <sip:sip-lab@127.0.0.1:5092>
Allow: MESSAGE, SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK
Supported: replaces, 100rel
Content-Length:  0


------------------------------------------------------------
2022/01/06 takeshi:

ChatGPT test:

Write sample code creating SDP for MRCP using pjmedia.

Here is an example of how you could create an SDP (Session Description Protocol) for MRCP (Media Resource Control Protocol) using PJSIP and PJMedia:

```
#include <pjmedia.h>
#include <pjmedia-codec.h>
#include <pjlib-util.h>
#include <pjlib.h>

/* Create the SDP for MRCP */
pj_status_t create_mrcp_sdp(pjmedia_sdp_session **sdp)
{
    pj_pool_t *pool;
    pjmedia_sdp_media *m;
    pjmedia_sdp_attr *attr;
    pj_status_t status;

    /* Create a pool for the SDP */
    pool = pj_pool_create(NULL, "sdp%p", 1024, 1024, NULL);
    if (!pool) {
        return PJ_ENOMEM;
    }

    /* Create the SDP session */
    *sdp = pj_pool_zalloc(pool, sizeof(pjmedia_sdp_session));
    if (!*sdp) {
        pj_pool_release(pool);
        return PJ_ENOMEM;
    }

    /* Set the SDP version */
    (*sdp)->origin.version = 0;

    /* Set the SDP origin */
    pj_strdup2((*sdp)->pool, &(*sdp)->origin.user, "mrcp");
    pj_strdup2((*sdp)->pool, &(*sdp)->origin.addr, "127.0.0.1");

    /* Set the SDP session name */
    pj_strdup2((*sdp)->pool, &(*sdp)->name, "MRCP Session");

    /* Create the SDP media */
    m = pj_pool_zalloc(pool, sizeof(pjmedia_sdp_media));
    if (!m) {
        pj_pool_release(pool);
        return PJ_ENOMEM;
    }

    /* Set the media type and port */
    pj_strdup2(pool, &m->desc.media, "audio");
    m->desc.port = 0;

    /* Set the media transport */
    pj_strdup2(pool, &m->desc.transport, "RTP/AVP");

    /* Add the media to the SDP */
    status = pjmedia_sdp_media_add_attr(m, &attr);
    if (status != PJ_SUCCESS) {
        pj_pool_release(pool);
        return status;
    }

    attr->name = pj_str("rtpmap");
    attr->value = pj_str("0 pcmu/8000");

    pjmedia_sdp_media_add_attr(m, &attr);
    if (status != PJ_SUCCESS) {
        pj_pool_release(pool);
        return status;
    }

    attr->name = pj_str("fmtp");
    attr->value = pj_str("0-15");

    pjmedia_sdp_session_add_media(sdp, m);

    /* Return success */
    return PJ_SUCCESS;
}
```

But we should have media for application TCP/MRCPv2 in the SDP.

Yes, you are correct. If you want to include a media description for the MRCPv2 application over TCP in the SDP, you can modify the create_mrcp_sdp function as follows:

```

pj_status_t create_mrcp_sdp(pjmedia_sdp_session **sdp)
{
    pj_pool_t *pool;
    pjmedia_sdp_media *m;
    pjmedia_sdp_attr *attr;
    pj_status_t status;

    /* Create a pool for the SDP */
    pool = pj_pool_create(NULL, "sdp%p", 1024, 1024, NULL);
    if (!pool) {
        return PJ_ENOMEM;
    }

    /* Create the SDP session */
    *sdp = pj_pool_zalloc(pool, sizeof(pjmedia_sdp_session));
    if (!*sdp) {
        pj_pool_release(pool);
        return PJ_ENOMEM;
    }

    /* Set the SDP version */
    (*sdp)->origin.version = 0;

    /* Set the SDP origin */
    pj_strdup2((*sdp)->pool, &(*sdp)->origin.user, "mrcp");
    pj_strdup2((*sdp)->pool, &(*sdp)->origin.addr, "127.0.0.1");

    /* Set the SDP session name */
    pj_strdup2((*sdp)->pool, &(*sdp)->name, "MRCP Session");

    /* Create the SDP media for the MRCPv2 application over TCP */
    m = pj_pool_zalloc(pool, sizeof(pjmedia_sdp_media));
    if (!m) {
        pj_pool_release(pool);
        return PJ_ENOMEM;
    }

    /* Set the media type and port */
    pj_strdup2(pool, &m->desc.media, "application");
    m->desc.port = 0;

    /* Set the media transport */
    pj_strdup2(pool, &m->desc.transport, "TCP/MRCPv2");

    /* Add the media to the SDP */
    status = pjmedia_sdp_media_add_attr(m, &attr);
    if (status != PJ_SUCCESS) {
        pj_pool_release(pool);
        return status;
    }

    /* Add any additional attributes as needed */
    /* ... */

    pjmedia_sdp_session_add_media(sdp, m);

    /* Return success */
    return PJ_SUCCESS;
}

```

This code creates a media object for the MRCPv2 application over TCP and sets its type, port, and transport. It then adds the media object to the SDP session. You can also add any additional attributes to the media object as needed.


Then I asked:

Write a function to create an SDP for MSRP using pjmedia.

Here is an example of how you could create an SDP (Session Description Protocol) for MSRP (Message Session Relay Protocol) using PJSIP and PJMedia:

```
#include <pjmedia.h>
#include <pjmedia-codec.h>
#include <pjlib-util.h>
#include <pjlib.h>

/* Create the SDP for MSRP */
pj_status_t create_msrp_sdp(pjmedia_sdp_session **sdp)
{
    pj_pool_t *pool;
    pjmedia_sdp_media *m;
    pjmedia_sdp_attr *attr;
    pj_status_t status;

    /* Create a pool for the SDP */
    pool = pj_pool_create(NULL, "sdp%p", 1024, 1024, NULL);
    if (!pool) {
        return PJ_ENOMEM;
    }

    /* Create the SDP session */
    *sdp = pj_pool_zalloc(pool, sizeof(pjmedia_sdp_session));
    if (!*sdp) {
        pj_pool_release(pool);
        return PJ_ENOMEM;
    }

    /* Set the SDP version */
    (*sdp)->origin.version = 0;

    /* Set the SDP origin */
    pj_strdup2((*sdp)->pool, &(*sdp)->origin.user, "msrp");
    pj_strdup2((*sdp)->pool, &(*sdp)->origin.addr, "127.0.0.1");

    /* Set the SDP session name */
    pj_strdup2((*sdp)->pool, &(*sdp)->name, "MSRP Session");

    /* Create the SDP media */
    m = pj_pool_zalloc(pool, sizeof(pjmedia_sdp_media));
    if (!m) {
        pj_pool_release(pool);
        return PJ_ENOMEM;
    }

    /* Set the media type and port */
    pj_strdup2(pool, &m->desc.media, "message");
    m->desc.port = 0;

    /* Set the media transport */
    pj_strdup2(pool, &m->desc.transport, "TCP/MSRP");

    /* Add the media to the SDP */
    status = pjmedia_sdp_media_add_attr(m, &attr);
    if (status != PJ_SUCCESS) {
        pj_pool_release(pool);
        return status;
    }

    /* Add any additional attributes as needed */
    /* ... */

    pjmedia_sdp_session_add_media(sdp, m);

    /* Return success */
    return PJ_SUCCESS;
}


```
This code creates a pool for the SDP, and then creates the SDP session and sets its version, origin, and name. Next, it creates a media object for the SDP and sets its type, port, and transport. Finally, it adds the media object to the SDP session. You can also add any additional attributes to the

This code is missing the part where media is set for message TCP/MSRP.

I apologize for the oversight in my previous response. Here is the corrected version of the create_msrp_sdp function that includes the media description for the MSRP message over TCP:

```
pj_status_t create_msrp_sdp(pjmedia_sdp_session **sdp)
{
    pj_pool_t *pool;
    pjmedia_sdp_media *m;
    pjmedia_sdp_attr *attr;
    pj_status_t status;

    /* Create a pool for the SDP */
    pool = pj_pool_create(NULL, "sdp%p", 1024, 1024, NULL);
    if (!pool) {
        return PJ_ENOMEM;
    }

    /* Create the SDP session */
    *sdp = pj_pool_zalloc(pool, sizeof(pjmedia_sdp_session));
    if (!*sdp) {
        pj_pool_release(pool);
        return PJ_ENOMEM;
    }

    /* Set the SDP version */
    (*sdp)->origin.version = 0;

    /* Set the SDP origin */
    pj_strdup2((*sdp)->pool, &(*sdp)->origin.user, "msrp");
    pj_strdup2((*sdp)->pool, &(*sdp)->origin.addr, "127.0.0.1");

    /* Set the SDP session name */
    pj_strdup2((*sdp)->pool, &(*sdp)->name, "MSRP Session");

    /* Create the SDP media for the MSRP message over TCP */
    m = pj_pool_zalloc(pool, sizeof(pjmedia_sdp_media));
    if (!m) {
        pj_pool_release(pool);
        return PJ_ENOMEM;
    }

    /* Set the media type and port */
    pj_strdup2(pool, &m->desc.media, "message");
    m->desc.port = 0;

    /* Set the media transport */
    pj_strdup2(pool, &m->desc.transport, "TCP/MSRP");

    /* Add the media to the SDP */
    status = pjmedia_sdp_media_add_attr(m, &attr);
    if (status != PJ_SUCCESS) {
        pj_pool_release(pool);
        return status;
    }

    /* Add any additional attributes as needed */
    /* ... */

    pjmedia_sdp_session_add_media(sdp, m);

    /* Return success */
    return PJ_SUCCESS;
}

```

This code creates a media object for the MSRP message over TCP and sets its type, port, and transport. It then adds the media object to the SDP session. You can also add any additional attributes to the media object as needed.

I hope this helps! Let me know if you have any questions.

------------------------------------------------------------

I am considering using pj_list to manage multiple media in the SIP calls.

So I asked ChatGPT:

How to use pj_list with pool?

The pj_list is a doubly-linked list implementation provided by the pjlib library. You can use a pj_list with a pj_pool_t memory pool by allocating memory for the list nodes from the pool.

Here is an example of how to use a pj_list with a pj_pool_t memory pool:
```

#include <pjlib.h>

int main()
{
    pj_pool_t *pool;
    pj_list_t list;

    // Create a memory pool.
    pool = pj_pool_create(NULL, "mypool", 1024, 1024, NULL);

    // Initialize the list.
    pj_list_init(&list);

    // Allocate a list node from the pool.
    pj_list_node *node = (pj_list_node*) pj_pool_alloc(pool, sizeof(pj_list_node));

    // Insert the node into the list.
    pj_list_insert_before(&list, node);

    // The node will be automatically released back to the pool when the pool is destroyed.
}

```

Note that when you allocate memory for a pj_list_node from a pj_pool_t, the memory will be automatically released back to the pool when the pool is destroyed. This can be convenient for managing the memory used by the list.

You can also use the pj_list_erase function to remove a node from the list and release its memory back to the pool. Here is an example:

```

#include <pjlib.h>

int main()
{
    pj_pool_t *pool;
    pj_list_t list;

    // Create a memory pool.
    pool = pj_pool_create(NULL, "mypool", 1024, 1024, NULL);

    // Initialize the list.
    pj_list_init(&list);

    // Allocate a list node from the pool.
    pj_list_node *node = (pj_list_node*) pj_pool_alloc(pool, sizeof(pj_list_node));

    // Insert the node into the list.
    pj_list_insert_before(&list, node);

    // Remove the node from the list and release its memory back to the pool.
    pj_list_erase(node);
}

```

But I could also use an array:

How to use pj_array_insert with pool?

You can use pj_array_insert with a pj_pool_t memory pool by allocating memory for the elements of the array from the pool.

Here is an example of how to use pj_array_insert with a pj_pool_t memory pool:
```

#include <pjlib.h>

int main()
{
    pj_pool_t *pool;
    pj_array_t array;

    // Create a memory pool.
    pool = pj_pool_create(NULL, "mypool", 1024, 1024, NULL);

    // Initialize the array with capacity for 10 elements.
    pj_array_init(pool, &array, 10, sizeof(int), NULL);

    // Allocate memory for an element from the pool.
    int *element = (int*) pj_pool_alloc(pool, sizeof(int));
    *element = 5;

    // Insert the element into the array.
    pj_array_insert(&array, element);

    // The element will be automatically released back to the pool when the pool is destroyed.
}
```

Note that when you allocate memory for an element of the array from a pj_pool_t, the memory will be automatically released back to the pool when the pool is destroyed. This can be convenient for managing the memory used by the array.

You can also use the pj_array_erase function to remove an element from the array and release its memory back to the pool. Here is an example:

```

#include <pjlib.h>

int main()
{
    pj_pool_t *pool;
    pj_array_t array;

    // Create a memory pool.
    pool = pj_pool_create(NULL, "mypool", 1024, 1024, NULL);

    // Initialize the array with capacity for 10 elements.
    pj_array_init(pool, &array, 10, sizeof(int), NULL);

    // Allocate memory for an element from the pool.
    int *element = (int*) pj_pool_alloc(pool, sizeof(int));
    *element = 5;

    // Insert the element into the array.
    pj_array_insert(&array, element);

    // Remove the element from the array and release its memory back to the pool.
    pj_array_erase(&array, element);
}

```
It is OK to use pj_array_insert when the capacity of the array was already reached?

Yes, it is okay to use pj_array_insert even if the capacity of the array has already been reached. When the capacity of the array has been reached and you call pj_array_insert, pj_array will automatically allocate more memory to store the new element.

For example:

```

#include <pjlib.h>

int main()
{
    pj_array_t array;
    int element;

    // Initialize the array with capacity for 1 element.
    pj_array_init(NULL, &array, 1, sizeof(int), NULL);

    // Insert an element into the array.
    element = 5;
    pj_array_insert(&array, &element);

    // Insert another element into the array.
    element = 6;
    pj_array_insert(&array, &element);

    // The array will automatically allocate more memory to store the second element.
}

```

Keep in mind that constantly allocating more memory to store new elements can be inefficient, especially if you are inserting a large number of elements into the array. In such cases, it may be more efficient to initialize the array with a larger capacity to begin with, to avoid the need for frequent reallocations.

You can use the pj_array_set_capacity function to set the capacity of the array to a specific value. This can be useful if you know in advance how many elements you will be inserting into the array and want to avoid the overhead of frequent reallocations.

Please show me code where the array itself is allocated from the pool.

Certainly. Here is an example of how you can allocate the pj_array_t structure itself from a pj_pool_t memory pool:

```

#include <pjlib.h>

int main()
{
    pj_pool_t *pool;
    pj_array_t *array;

    // Create a memory pool.
    pool = pj_pool_create(NULL, "mypool", 1024, 1024, NULL);

    // Allocate the array structure from the pool.
    array = (pj_array_t*) pj_pool_alloc(pool, sizeof(pj_array_t));

    // Initialize the array with capacity for 10 elements.
    pj_array_init(pool, array, 10, sizeof(int), NULL);

    // Insert an element into the array.
    int element = 5;
    pj_array_insert(array, &element);

    // The array and its elements will be automatically released back to the pool when the pool is destroyed.
}

```

In this example, the pj_array_t structure is allocated from the memory pool and the array is initialized with a capacity for 10 elements. When you insert an element into the array, pj_array will allocate memory for the element from the pool. When the pool is destroyed, the memory used by the array and its elements will be automatically released back to the pool.

Keep in mind that when you allocate the pj_array_t structure from a pj_pool_t, you must also pass the memory pool to the pj_array_init function as the first argument. This will ensure that the array allocates memory for its elements from the specified pool.


------------------------------------------------------------
2023/01/08 takeshi:

Here are some samples of different media session offers and answers:

Audio:
```
INVITE sip:UserB2@ there.com SIP/2.0
Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2
Via: SIP/2.0/UDP here.com:5060
Record-Route:
From: BigGuy
To: LittleGuy
Call-ID: 12345600@here.com
CSeq: 1 INVITE
Contact: BigGuy
Content-Type: application/sdp
Content-Length: ...

v=0
o=UserA 2890844526 2890844526 IN IP4 client.here.com
s=Session SDP
c=IN IP4 100.101.102.103
t=0 0
m=audio 49170 RTP/AVP 0
a=rtpmap:0 PCMU/8000


SIP/2.0 200 OK
Via: SIP/2.0/UDP wcom.com:5060; branch=2
Via: SIP/2.0/UDP here.com:5060
Record-Route:
From: TheBigGuy
To: TheLittleGuy ;tag=123456
Call-Id: 12345600@here.com
CSeq: 1 INVITE
Contact: TheLittleGuyVoiceMail
Content-Type: application/sdp
Content-Length:

v=0
o=UserB 2890844527 2890844527 IN IP4 vm.wcom.com
s=Session SDP
c=IN IP4 110.111.112.114
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000 

```

Audio and Video:
```

INVITE sip:68854109@36.13.226.123:5330 SIP/2.0
Via: SIP/2.0/UDP 202.173.5.227;branch=z9hG4bK0ad6.98f64bc6fb1902d8a641ecf938033a0e.0
Via: SIP/2.0/UDP 192.168.50.101:5791;received=36.13.226.123;branch=z9hG4bKLXpxn8j6AvZU87ke;rport=5791
Contact: <sip:14084112@36.13.226.123:5791>
Max-Forwards: 69
From: <sip:05068608167@softphone.spc.brastel.ne.jp>;tag=E03BBCA8FF01003D4CF07357422B93FE
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces, path
Content-Type: application/sdp
To: "05068634601" <sip:68854109@softphone.spc.brastel.ne.jp>;tag=A96401AB587CE38E37AD7DC61D6564A1
Call-ID: 66E9237FD24ED124E4647BBDB9AC0EBF853FFAB1
CSeq: 1 INVITE
User-Agent: Softphone/5.9.36 (build 1783094; iOS 15.6; arm64-neon)
Content-Length: 546

v=0
o=- 0924519482 25587 IN IP4 172.26.170.170
s=unazhle
c=IN IP4 202.173.5.228
t=0 0
m=audio 35900 RTP/AVP 3 0 8 18 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=rtcp:35901
a=ptime:20
m=video 35916 RTP/AVP 99 108
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=42800d;packetization-mode=1;level-asymmetry-allowed=1
a=rtpmap:108 VP8/90000
a=fmtp:108 max-fr=30;max-fs=3600
a=sendrecv
a=rtcp:35917


SIP/2.0 200 OK
Via: SIP/2.0/UDP 202.173.5.227;branch=z9hG4bK0ad6.98f64bc6fb1902d8a641ecf938033a0e.0
Via: SIP/2.0/UDP 192.168.50.101:5791;received=36.13.226.123;branch=z9hG4bKLXpxn8j6AvZU87ke;rport=5791
Record-Route: <sip:202.173.5.227;lr=on;ftag=A96401AB587CE38E37AD7DC61D6564A1;vsf=AAAAAAYNCAMMBwMNdkNeJgcfDhseDUEdFU1dEhFPEQYEH1oLCQAEFS5qcA--;did=96d.253>
Contact: <sip:68854109@192.168.50.236:5330>
From: <sip:05068608167@softphone.spc.brastel.ne.jp>;tag=E03BBCA8FF01003D4CF07357422B93FE
Call-ID: 66E9237FD24ED124E4647BBDB9AC0EBF853FFAB1
CSeq: 1 INVITE
To: "05068634601" <sip:68854109@softphone.spc.brastel.ne.jp>;tag=A96401AB587CE38E37AD7DC61D6564A1
Allow: ACK, BYE, CANCEL, INFO, INVITE, MESSAGE, NOTIFY, OPTIONS, REFER
Supported: replaces, path
Content-Type: application/sdp
User-Agent: Softphone/5.6.5 (build 1796064; Android 9; arm64-v8a)
Content-Length: 341

v=0
o=- 2720741187 48846 IN IP4 172.26.170.170
s=bngitxz
c=IN IP4 192.168.50.236
t=0 0
m=audio 60288 RTP/AVP 3 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
m=video 52406 RTP/AVP 99
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=42800d;packetization-mode=1;level-asymmetry-allowed=1
a=sendrecv
```


T.38:
```
INVITE sip:+1-650-555-2222@iftgw.there.com;user=phone SIP/2.0 
Via: SIP/2.0/UDP obelix.wcom.com:5060; branch=2d007.1 
Via: SIP/2.0/UDP ifax.here.com:5060 
Record-Route: <sip:+1-650-555-2222@iftgw.there.com;  
             maddr=obelix.wcom.com> 
From: sip:+1-303-555-1111@ifax.here.com;user=phone;tag=ab111 
To: sip:+1-650-555-2222@obelix.wcom.com;user=phone 
Call-ID: 1717@ifax.here.com 
CSeq: 17 INVITE 
Contact: <sip:+1-303-555-1111@ifax.here.com> 
Content-Type: application/sdp 
Content-Length: 320 

v=0 
o=ifaxgw1 2890846527 2890846527 IN IP4 ifax.here.com 
s=Session SDP 
c=IN IP4 ifaxmg.here.com 
t=0 0 
m=image 15002 udptl t38 
a=T38FaxVersion:0 
a=T38MaxBitRate:14400 
a=T38FaxFillBitRemoval:0 
a=T38FaxTranscodingMMR:0 
a=T38FaxTranscodingJBIG:0 
a=T38FaxRateManagement:transferredTCF 
a=T38FaxMaxBuffer:72 
a=T38FaxMaxDatagram:316 
a=T38FaxUdpEC:t38UDPFEC 
a=T38FaxUdpEC:t38UDPRedundancy 


SIP/2.0 200 OK 
Via: SIP/2.0/UDP obelix.wcom.com:5060; branch=2d007.1 
Via: SIP/2.0/UDP ifax.here.com:5060 
From: sip:+1-303-555-1111@ifax.here.com;user=phone;tag=ab11 
To: sip:+1-650-555-2222@obelix.wcom.com;user=phone;tag=cde17 
Call-ID: 1717@ifax.here.com 
CSeq: 17 INVITE 
Contact: <sip:+1-650-555-2222@iftgw.there.com> 
Content-Type: application/sdp 
Content-Length: 320 

v=0 
o=faxgw1 2890844527 2890844527 IN IP4 iftgw.there.com 
s=Session SDP 
c=IN IP4 iftmg.there.com 
t=0 0 
m=image 49172 udptl t38 
a=T38FaxVersion:0 
a=T38MaxBitRate:14400
```

MRCPv2:
```
INVITE sip:mresources@example.com SIP/2.0
Via:SIP/2.0/TCP client.atlanta.example.com:5060;
branch=z9hG4bK74bf9
Max-Forwards:6
To:MediaServer <sip:mresources@example.com>
From:sarvi <sip:sarvi@example.com>;tag=1928301774
Call-ID:a84b4c76e66710
CSeq:314161 INVITE
Contact:<sip:sarvi@client.example.com>
Content-Type:application/sdp
Content-Length:...

v=0
o=sarvi 2890844526 2890842808 IN IP4 192.0.2.4
s=-
c=IN IP4 192.0.2.12
m=application 9 TCP/MRCPv2 1
a=setup:active
a=connection:new
a=resource:speechsynth
a=cmid:1
m=audio 49170 RTP/AVP 0
a=rtpmap:0 pcmu/8000
a=recvonly
a=mid:1


SIP/2.0 200 OK
Via:SIP/2.0/TCP client.atlanta.example.com:5060;
branch=z9hG4bK74bf9
To:MediaServer <sip:mresources@example.com>;tag=62784
From:sarvi <sip:sarvi@example.com>;tag=1928301774
Call-ID:a84b4c76e66710
CSeq:314161 INVITE
Contact:<sip:mresources@server.example.com>
Content-Type:application/sdp
Content-Length:...

v=0
o=- 2890844526 2890842808 IN IP4 192.0.2.4
s=-
c=IN IP4 192.0.2.11
m=application 32416 TCP/MRCPv2 1
a=setup:passive
a=connection:new
a=channel:32AECB234338@speechsynth
a=cmid:1
m=audio 48260 RTP/AVP 0
a=rtpmap:0 pcmu/8000
a=sendonly
a=mid:1
```

MSRP (some header lines removed for brevity):
```
INVITE sip:bob@biloxi.example.com SIP/2.0
To: <sip:bob@biloxi.example.com>
From: <sip:alice@atlanta.example.com>;tag=786
Call-ID: 3413an89KU
Content-Type: application/sdp

c=IN IP4 atlanta.example.com
m=message 7654 TCP/MSRP *
a=accept-types:text/plain
a=path:msrp://atlanta.example.com:7654/jshA7weztas;tcp


SIP/2.0 200 OK
To: <sip:bob@biloxi.example.com>;tag=087js
From: <sip:alice@atlanta.example.com>;tag=786
Call-ID: 3413an89KU
Content-Type: application/sdp

c=IN IP4 biloxi.example.com
m=message 12763 TCP/MSRP *
a=accept-types:text/plain
a=path:msrp://biloxi.example.com:12763/kjhd37s2s20w2a;tcp
```

Checking just the m= lines:

```
Audio:

m=audio 49170 RTP/AVP 0

m=audio 3456 RTP/AVP 0


Video:

m=audio 35900 RTP/AVP 3 0 8 18 101
m=video 35916 RTP/AVP 99 108

m=audio 60288 RTP/AVP 3 101
m=video 52406 RTP/AVP 99


T.38:

m=image 15002 udptl t38 

m=image 49172 udptl t38 


MRCP:

m=application 9 TCP/MRCPv2 1
m=audio 49170 RTP/AVP 0

m=application 32416 TCP/MRCPv2 1
m=audio 48260 RTP/AVP 0


MSRP:

m=message 7654 TCP/MSRP *

m=message 12763 TCP/MSRP *
```

So here is a sample of new optional parameter media_endpoints to be passed to functions that send SDP offer/answer messages:
```
"media_endpoints": [
	{
		"type": "audio"
	},
	{
		"type": "video"
	},
	{
		"type": "t38",
		"attributes": [
			"T38FaxVersion:0",
			"T38MaxBitRate:14400",
			"T38FaxFillBitRemoval:0",
			"T38FaxTranscodingMMR:0",
			"T38FaxTranscodingJBIG:0"
		]
	},
	{
		"type": "mrcp",
		"attributes": ["resource:speechsynth", "recvonly"]
	},
	{
		"type": "msrp",
		"attributes": ["accept-types:text/plain"]
	}
]
```

For audio and video, for start there will be no attributes as they will be generated based on the codecs enabled in pjmedia (we might add support for audio/video attributes later)

So when this is not passed it will be assumed to be:
```
"media_endpoints": [
    {
        "type": "audio"
    }
]

and so sip commands in existing scripts will not be affected.

Obs: instead of "media_endpoints" we could have used "media_streams" however MRCP and MSRP are not stream oriented media (they are connection oriented media as the use TCP).
That's why we decided to use media_endpoints.

----------------------------------------------------
2023/01/08 takeshi:

Instead of pjmedia_endpt_create_sdp() that would create the full SDP for use including the m= and a= lines, we are going to use pjmedia_endpt_create_audio_sdp() so that it can be invoked for each audio endpoint.
However, because of this, we will need to use
  pjmedia_sdp_attr_create()
and
  pjmedia_sdp_attr_add()
to add the attributes for the media sections (that start with the m= lines)

Here is a typical SDP currently generated by sip-lab:

v=0
o=- 3882147696 3882147696 IN IP4 127.0.0.1
s=pjmedia
t=0 0
m=audio 10000 RTP/AVP 0 8 3 96 97 98 99 9 18 120 121 122
c=IN IP4 127.0.0.1
b=TIAS:64000
a=rtcp:10001 IN IP4 127.0.0.1
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:96 iLBC/8000
a=fmtp:96 mode=30
a=rtpmap:97 speex/32000
a=rtpmap:98 speex/16000
a=rtpmap:99 speex/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=rtpmap:120 telephone-event/8000
a=fmtp:120 0-16
a=rtpmap:121 telephone-event/32000
a=fmtp:121 0-16
a=rtpmap:122 telephone-event/16000
a=fmtp:122 0-16


Also, here sample SDP for SRTP:

v=0
o=jdoe 2890844526 2890842807 IN IP4 10.47.16.5
s=SDP Seminar
i=A Seminar on the session description protocol
u=http://www.example.com/seminars/sdp.pdf
e=j.doe@example.com (Jane Doe)
c=IN IP4 161.44.17.12/127
t=2873397496 2873404696
m=video 51372 RTP/SAVP 31
a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:d0RmdmcmVCspeEc3QGZiNWpVLFJhQX1cfHAwJSoj|2^20|1:32
m=audio 49170 RTP/SAVP 0
a=crypto:1 AES_CM_128_HMAC_SHA1_32
inline:NzB4d1BINUAvLEw6UzF3WSJ+PSdFcGdUJShpX1Zj|2^20|1:32
m=application 32416 udp wb
a=orient:portrait

https://www.rfc-editor.org/rfc/rfc4568.html

--------------------------------------------------------
2023/01/08 takeshi:

I asked ChatGPT to show we code to answer a call and specifying the media format that I accected:


Here is an example of how you can answer a call and specify the media format that you accept using the PJSIP and PJMEDIA libraries:

```
#include <pjsip.h>
#include <pjsip_ua.h>
#include <pjmedia.h>
#include <pjmedia-codec.h>

// ...

// Callback function for incoming calls
static pj_bool_t on_incoming_call(pjsua_acc_id acc_id, pjsua_call_id call_id,
                                   pjsip_rx_data *rdata)
{
    // Create a SDP session description for the call
    pjmedia_sdp_session *sdp;
    pjmedia_sdp_neg *neg;
    pj_status_t status;

    // Get the codec ID for the media format that we accept
    pj_str_t codec_id = pj_str("the_codec_id");

    // Create an SDP session description with one media stream
    // using the codec that we accept
    status = pjmedia_endpt_create_sdp(pjsua_get_pjmedia_endpt(), rdata->tp_info.pool, 1,
                                      &codec_id, &sdp);
    if (status != PJ_SUCCESS) {
        // error handling
        return PJ_FALSE;
    }

    // Create an SDP negotiator and set the local and remote SDP
    // session descriptions
    status = pjmedia_sdp_neg_create_w_remote_offer(rdata->tp_info.pool, sdp, &neg);
    if (status != PJ_SUCCESS) {
        // error handling
        return PJ_FALSE;
    }
    pjmedia_sdp_neg_set_remote_offer(neg, rdata->msg_info.msg->body->data);

    // Answer the call with the SDP negotiator
    pjsip_tx_data *tdata;
    status = pjsip_inv_answer(pjsua_call_get_invite(call_id), 200, NULL, neg, &tdata);
    if (status != PJ_SUCCESS) {
        // error handling
        return PJ_FALSE;
    }

    // Send the response
    status = pjsip_inv_send_msg(pjsua_call_get_invite(call_id), tdata);
    if (status != PJ_SUCCESS) {
        // error handling
        return PJ_FALSE;
    }

    return PJ_TRUE;
}
```

In this example, the on_incoming_call() callback function is called when an incoming call is received. It creates an SDP session description with one media stream using the codec that you accept, and then creates an SDP negotiator and sets the local and remote SDP session descriptions. Finally, it answers the call using the pjsip_inv_answer() function and the SDP negotiator.

-----------------------------------------------------
2023/01/09 takeshi:

I was working on #42 following:
  https://www.pjsip.org/pjmedia/docs/html/group__PJMEDIA__SESSION.htm#ga4b9d005c55d892a13a7dd49a9167b974

but pjmedia_session was deprecated long ago:
  https://github.com/pjsip/pjproject/issues/1207


-----------------------------------------------------
2023/01/09 takeshi:

Basic changes for #42 were completed:
  - works with reinvite
  - works with delayed media (including reinvites with delayed media)
  - sends full list of codecs (solves #3).
But only SIP works.
We need to update the code to create/start/recreate the media streams upon SDP negotiation.
Then after this gets working we will add code to support connection-based (TCP/TLS) media (MRCP, MSRP etc).

Draft:
  - add call->media_neg and call->media_neg_count
  - in call_create, call_respond(with_media) call_reinvite, we populate the call->media_neg array (creating media components as necessary)
  - when building call->media_nego we must search for the same media already present in call->media. 
    - If the media is already present, copy the pointer from call->media to media->neg (the search should be done igonoring specifics like port number, we just need to search for the same type).
    - else create a media endpoint anew and add it to media->neg
  - always use call->media_nego to create local_sdp to be sent to the peer
  - at on_media_update:
    - if status==PJ_SUCCESS (negotiation successful)
      - then
        - get the local active_sdp
        - for each m in call->media_neg:
          - if (m matches a media at active_sdp) and m is not present in call->media, move the m to call->media (which is the array with active media).
          - else remove m from call->media_neg (destroy media components if necessary)
          - after this operation, call->media_neg should be empty
        - for each m in call->media:
          - if (m doesn't match a media at active_sdp) then remove m from call->media (destroy media components if necessary) 
      - else remove all m from call->media_neg (destroy media components if necessary)

UPDATE: the above was implemented but not tested yet.

After it confirms to be working then we should add code to check the media streams and create/destroy as needed.

-----------------------------------------------------
2023/01/12 takeshi:

Working on #42.

Regarding event 'media_status':
        {
            event: 'media_status',
            call_id: oc.id,
            status: 'setup_ok',
            local_mode: 'sendrecv',
            remote_mode: 'sendrecv',
        },
it would need to be modified to report a list of media instead of a single media.

But I am thinking in eliminating this event as now that we report both REINVITEs and their replies, if we want to check the status, we could just check the SDP.

This could be done either by a string-match for the whole body with data capture and with irrelevant and variable data being ignored by using !{_}.
However, it would be good to have sdp-parsing and sdp-matching modules to permit to easily match specific elements in the SDP.

Maybe we could get inspiration from XML xpath or jq.
So we would convert the SDP to a JSON object and use queries to indicate the elements we are interested in.

Here is a sample SDP:

v=0
o=- 3882513420 3882513420 IN IP4 0.0.0.0
s=pjmedia
t=0 0
m=application 9 TCP/MRCPv2 1
c=IN IP4 127.0.0.1
m=audio 10000 RTP/AVP 0 8 3
c=IN IP4 127.0.0.1
b=TIAS:64000
a=rtcp:10001 IN IP4 127.0.0.1
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=sendrecv


Notice that we have keys before the m= lines.
These are session parameters and could be specified just like:
    {
        v: '0',
        o: '- 3882513420 3882513420 IN IP4 0.0.0.0',
        s: 'pjmedia',
        t: '0 0',
    }

Then for the media elements we could do it like this:
    {
        m0m: 'application 9 TCP/MRCPv2 1',
        m0c: 'IN IP4 127.0.0.1',
        m1m: 'audio 10000 RTP/AVP 0 8 3',
        m1c: 'IN IP4 127.0.0.1',
        m1b: 'TIAS:64000',
        m1a0: 'rtcp:10001 IN IP4 127.0.0.1',
        m1a1: 'sendrecv',
    }

So m1a1 means: media 1, attribute 1.

Also we could use some wildcards like:
    {
        m_m: 'audio 10000 RTP/AVP 0 8 3',
    }

which would match any m0m, m1m, m2m etc.
And:
    {
        m_a_: 'rtcp:10001 IN IP4 127.0.0.1',
    }
would match any m0a0, m0a1, m0a2, m1a0, m1a1, m1a2 etc.

Obs: if media setup fails we would raise an error event for the channel.
This would make scripts easier to write.

UPDATE: thinking again, we will create mrcp-matching that will convert the body to an object. Then we will use jq to select items to match.

Research:

takeshi@takeshi-desktop:~/tmp$ cat b.json | jq .
{
  "media": [
    {
      "type": "audio",
      "attrs": [
        "rtpmap:1234",
        "sendrecv"
      ]
    },
    {
      "type": "video",
      "attrs": [
        "rtpmap:4321",
      ]
    }
  ]
}

takeshi@takeshi-desktop:~/tmp$ cat b.json | jq '.media[] | select(.type=="audio").attrs'
[
  "rtpmap:1234",
  "sendrecv"
]

takeshi@takeshi-desktop:~/tmp$ cat b.json | jq '.media[] | select(.type=="audio").attrs | any(. == "sendonly")'
false

takeshi@takeshi-desktop:~/tmp$ cat b.json | jq '.media[] | select(.type=="audio").attrs | any(. == "sendrecv")'
true


So to check hold/unhold the key for element selection would need to be specified like this:

var expected = {
    '.media[] | select(.type=="audio").attrs | any(. == "sendrecv")': true,
}

But although this would be very exact, for cases where there is just one media, we can simply match against the body of the sip message:

var expected = {
    msg: sip_msg({
        $rb: '!{_}a:sendrecv'
    }),
}

-----------------------------------------------------
2023/02/05 takeshi

Regarding attribute sendrecv, sendonly, recvonly, inactive:


   If a stream is offered as sendonly, the corresponding stream MUST be
   marked as recvonly or inactive in the answer.  If a media stream is
   listed as recvonly in the offer, the answer MUST be marked as
   sendonly or inactive in the answer.  If an offered media stream is
   listed as sendrecv (or if there is no direction attribute at the
   media or session level, in which case the stream is sendrecv by
   default), the corresponding stream in the answer MAY be marked as
   sendonly, recvonly, sendrecv, or inactive.  If an offered media
   stream is listed as inactive, it MUST be marked as inactive in the
   answer.


https://www.ietf.org/rfc/rfc3264.txt


I think we should first work to have parameter mode (replacing hold) working on each AudioEndpoint (later we will allow a general mode parameter in the json options that could be overwritten by each AudioEndpoint)

Draft:
  - remove all code related to hold/unhold (DONE)

  - add 
      pjmedia_sdp_session *remote_sdp;
    to struct Call (DONE)

  - update on_rx_offer2 to:
    - save call->remote_sdp (DONE)
    - in case of delayed_media, compute the answer mode attribute (sendrecv, sendonly, recvonly, inactive) based on call->local_sdp and call->remote_sdp and send it.
      obs: since call->local_sdp was precomputed, we need to remove old mode attribute if necessary before adding the new attribute value

  - write function gen_mode_for_answer to compute the value of the mode for answer based on mode in offer.

  - write function 
      int get_media_stream_mode(pjmedia_sdp_media *media) 
    to get the mode of a stream

  - update calls to process_media (remove parameter hold and let each AudioEndpoint to generate the correct attribute based on local and remote sdp media):
    - update each AudioEndpoint in call->media according the value of media mode in the document
    - for each media:
       - if call->remote_media is not set, send the mode attr according to local media endpoints
       - else find the remote pjmedia_sdp_media and use get_media_stream_mode() and gen_mode_for_answer

However, let's check if pjmedia does adjustment of mode by itself based on what each side sets (if so, we will not need to handle mode in answer, just set it as specified in the document)

-----------------------------------------------------
2023/02/22 takeshi:

How hold is signaled in SIP:

According to RFC 6337 a user can hold calls by sending a new SDP offer in an established session (Re-INVITE on active call), with an SDP payload of a=sendonly for each media stream the user want’s to hold.

The SIP Switch / PBX / UAS replies with an updated SDP where each media stream’s SDP contains a=recvonly.

To retrieve the call another SIP re-invite is sent by the UAC, this time setting the media attribute back to sendrecv


https://nickvsnetworking.com/sip-hold-with-rfc6337/

-------------------------
2023/02/22 takeshi:

We replaced
  hold: true|false 
with
  mode: sendrecv|sendonly|recvonly|inactive
and corrected code to properly set the media attribute value based on the local mode.

Ref:

   If a stream is offered as sendonly, the corresponding stream MUST be
   marked as recvonly or inactive in the answer.  If a media stream is
   listed as recvonly in the offer, the answer MUST be marked as
   sendonly or inactive in the answer.  If an offered media stream is
   listed as sendrecv (or if there is no direction attribute at the
   media or session level, in which case the stream is sendrecv by
   default), the corresponding stream in the answer MAY be marked as
   sendonly, recvonly, sendrecv, or inactive.  If an offered media
   stream is listed as inactive, it MUST be marked as inactive in the
   answer.

https://www.ietf.org/rfc/rfc3264.txt

The mode can be set at the root of the JSON document or can be set individually per media that supports streaming (currently only the ones of type=audio).

-------------------------
2023/02/23 takeshi:

We created struct Request to permit to respond to non-dialog SIP requests (like REGISTER, OPTIONS, MESSAGE, INFO).

-------------------------
2023/02/23 takeshi:

There was an issue with notify being sent to itself by UAC (should have been sent by the UAS to the UAC):

              127.0.0.1:5090                127.0.0.1:5092  │Via: SIP/2.0/UDP 127.0.0.1:5090;rport;branch=z9hG4bKPjadf79a44-6914-4025-b1b9-f277aac2d4fd                                                    
          ──────────┬─────────          ──────────┬─────────│Max-Forwards: 70                                                                                                                              
  14:07:23.386208   │          SUBSCRIBE          │         │From: <sip:user1@test1.com>;tag=e7aa8818-1f78-4905-8b1b-009b63ec7e9e                                                                          
        +0.048537   │ ──────────────────────────> │         │To: <sip:user1@test1.com>                                                                                                                     
  14:07:23.434745   │           200 OK            │         │Contact: <sip:user1@127.0.0.1:5090>                                                                                                           
        +0.000157   │ <────────────────────────── │         │Call-ID: c596520a-2f8c-4efd-8dec-70348b125920                                                                                                 
  14:07:23.434902   │─┐ NOTIFY                    │         │CSeq: 25548 SUBSCRIBE                                                                                                                         
        +0.050352   │<┘                           │         │Route: <sip:127.0.0.1:5092;lr>                                                                                                                
  14:07:23.485254   │─┐ 200 OK                    │         │Event: dialog                                                                                                                                 
        +0.054625   │<┘                           │         │Expires: 120                                                                                                                                  
  14:07:23.539879   │─┐ NOTIFY                    │         │Supported: replaces, 100rel                                                                                                                   
        +0.046769   │<┘                           │         │Accept: application/dialog-info+xml                                                                                                           
  14:07:23.586648   │─┐ 200 OK                    │         │Allow-Events: refer, dialog                                                                                                                   
                    │<┘                           │         │Content-Length:  0                                                                                                                            
                    │                             │         │                                                                                                                                              
                    │                             │         │         

This was happening because we forgot to call dlg_set_transport() when processing the SUBSCRIBE request.

-------------------------
2023/02/23 takeshi:

pjw_call_refer was disabled.

Instead REFER will be done by pjw_call_send_request as any other in-dialog request aside from BYE and CANCEL.

-------------------------
2023/02/25 takeshi:

Adding support for MRCPv2.

We will need to store data till we get a complete MRCP message:

We can use a simple HTTP server as basis for it:
  https://web.archive.org/web/20140409134855/http://www.ibm.com/developerworks/systems/library/es-nweb/sidefile1.html
  https://web.archive.org/web/20140905115151/http://www.ibm.com/developerworks/systems/library/es-nweb/
  https://stackoverflow.com/questions/176409/build-a-simple-http-server-in-c

UPDATE: no, we will not use an HTTP server. Instead we will notify tcp_data event to node.js that then will accumulate and parse the data and generate events.

-------------------------
2023/02/26 takeshi:

To easily test alternative addons, have modified binding.gyp to have a base project named 'all-settings' that will be used by project 'addon' and eventually by other addons.
Ref:
  https://stackoverflow.com/questions/31202931/reuse-build-condition-for-multiple-targets-in-binding-gyp

-------------------------
2023/03/14 takeshi:

Regarding MRCPv2 support.
Let's start with the receiving part of the TCP socket to generate the tcp_data event.

So, when processing media, in case we are the client side, we should do nothing. But if we are the server, we should allocate a TCP socket to listen to.
Then once the media_update happens, if the mrcp offer is accepted, the client should connect to the server.

However, since a server TCP socket can receive connections from multiple clients, we should prepare it so that we only accept the first connection and close the listening TCP socket after that.
This way we don't need to add things like connection_id to events (irrelevant for our tests). 
Obs: we could try to test things like closing connections during a session to simulate issues but I think this is of minimal value and so I will not spend time supporting this (it could be done instead using node.js module sip-mrcp.

Regarding, sending data, we must allocate memory on the dialog pool as the data must be available till the send operation completes.

Implementation draft:
  - create function create_tcp_socket("call", "mrcp|msrp|etc", **selected_port). This should return the pj_activesock created or NULL. The function should call pj_activesock_set_user_data(asock, {sock, call, proto}) so that in the callbacks, we will have a reference to the data necessary to generate the event tcp_data.
  - when activesock accept completes, the asock should be closed (clear user_data first) and its user_data should be passed to the new asock. We also need to update the media_endpoint with reference to the new socket as we will need to close this new one.
  - when close_media_endpoint is called, call pj_activesock_set_user_data(asock, NULL) so that the callbacks will ignore further data. Then call pj_activesock_close() and this will also close the underlying socket.

-------------------------
2024/03/05 takeshi:

Work on branch conf_bridge_arch is almost complete (need to create pjmedia_fax_port).
Most of functional calls pass but a few are causing crashes. Ex:

on_state_changed

e->type=5
inv->state=6
call will terminate call=d030fd88

close_media call_id=0
close_media_endpoint d03103b0
close_media_transport d0347358
audio_endpoint_remove_port
node: ../src/pj/os_core_unix.c:1492: pj_mutex_unlock: Assertion `mutex->owner == pj_thread_this()' failed.
./rr: line 45: 203195 Aborted                 (core dumped) node $i
samples/srtp.js failed


Obs: the above is hard to reproduce. It takes a lot of tries.


And samples/100_calls.js get stuck this way:

17:41:38.111  All expected events received
17:41:38.111  wait (tt:93) finished
json_get_string_param digits
inband
json_get_string_param digits
inband
json_get_string_param digits
inband
json_get_string_param digits
inband


(after the above happens, it is not possible to stop the app with Ctrl-C).

So I executed it inside gdb and stopped the app with Control-C.
Then I got:

Thread 1 "node" received signal SIGINT, Interrupt.
futex_wait (private=0, expected=2, futex_word=0x7ffff4cb28b8) at ../sysdeps/nptl/futex-internal.h:146
146	../sysdeps/nptl/futex-internal.h: No such file or directory.
(gdb) bt
#0  futex_wait (private=0, expected=2, futex_word=0x7ffff4cb28b8) at ../sysdeps/nptl/futex-internal.h:146
#1  __GI___lll_lock_wait (futex=futex@entry=0x7ffff4cb28b8, private=0) at ./nptl/lowlevellock.c:49
#2  0x00007ffff789805d in lll_mutex_lock_optimized (mutex=0x7ffff4cb28b8) at ./nptl/pthread_mutex_lock.c:48
#3  ___pthread_mutex_lock (mutex=mutex@entry=0x7ffff4cb28b8) at ./nptl/pthread_mutex_lock.c:128
#4  0x00007ffff4f47d50 in pj_mutex_lock (mutex=0x7ffff4cb28b8) at ../src/pj/os_core_unix.c:1447
#5  0x00007ffff4ed4201 in pjmedia_conf_add_port (conf=0x7ffff7affb30, pool=0x7fffeddcf380, strm_port=0x7fffed87faa8, port_name=0x7fffed87faa8, p_port=0x7fffeddce040)
    at ../src/pjmedia/conference.c:793
#6  0x00007ffff4eaeab0 in prepare_tonegen (c=<optimized out>, c=<optimized out>, ae=0x7fffeddcddd8) at ../src/sip.cpp:6383
#7  prepare_tonegen (c=<optimized out>, ae=0x7fffeddcddd8) at ../src/sip.cpp:6365
#8  0x00007ffff4eaebf2 in audio_endpoint_send_dtmf (call=0x7fffedd90598, ae=0x7fffeddcddd8, digits=0x7fffffffb7e0 "4321", mode=<optimized out>) at ../src/sip.cpp:3011
#9  0x00007ffff4eaed0e in send_dtmf (call=call@entry=0x7fffedd90598, digits=0x7fffffffb7e0 "4321", mode=mode@entry=1) at ../src/sip.cpp:3047
#10 0x00007ffff4ebd469 in pjw_call_send_dtmf (call_id=call_id@entry=139, json=<optimized out>) at ../src/sip.cpp:3136
#11 0x00007ffff4eca5d9 in call_send_dtmf (info=...) at /usr/include/c++/11/bits/basic_string.h:194
#12 0x00007ffff4ecf885 in Napi::details::CallbackData<Napi::Value (*)(Napi::CallbackInfo const&), Napi::Value>::Wrapper(napi_env__*, napi_callback_info__*)::{lambda()#1}::operator()() const (__closure=<synthetic pointer>, __closure=<synthetic pointer>) at /home/takeshi/src/git/MayamaTakeshi/sip-lab/node_modules/node-addon-api/napi-inl.h:111
#13 Napi::details::WrapCallback<Napi::details::CallbackData<Napi::Value (*)(Napi::CallbackInfo const&), Napi::Value>::Wrapper(napi_env__*, napi_callback_info__*)::{lambda()#1}>(Napi::details::CallbackData<Napi::Value (*)(Napi::CallbackInfo const&), Napi::Value>::Wrapper(napi_env__*, napi_callback_info__*)::{lambda()#1}) (callback=...)
    at /home/takeshi/src/git/MayamaTakeshi/sip-lab/node_modules/node-addon-api/napi-inl.h:73
#14 Napi::details::CallbackData<Napi::Value (*)(Napi::CallbackInfo const&), Napi::Value>::Wrapper (env=0x7ffff7e6f540, info=0x7fffffffc9d0)
    at /home/takeshi/src/git/MayamaTakeshi/sip-lab/node_modules/node-addon-api/napi-inl.h:106
#15 0x0000000000b861f9 in v8impl::(anonymous namespace)::FunctionCallbackWrapper::Invoke(v8::FunctionCallbackInfo<v8::Value> const&) ()
#16 0x0000000000e2f2a0 in v8::internal::MaybeHandle<v8::internal::Object> v8::internal::(anonymous namespace)::HandleApiCallHelper<false>(v8::internal::Isolate*, v8::internal::Handle<v8::internal::HeapObject>, v8::internal::Handle<v8::internal::FunctionTemplateInfo>, v8::internal::Handle<v8::internal::Object>, unsigned long*, int) ()
#17 0x0000000000e306f6 in v8::internal::Builtin_HandleApiCall(int, unsigned long*, v8::internal::Isolate*) ()
#18 0x00000000018384f9 in Builtins_CEntry_Return1_DontSaveFPRegs_ArgvOnStack_BuiltinExit ()
#19 0x00000000017b06dc in Builtins_InterpreterEntryTrampoline ()
#20 0x000030e5ef9815b1 in ?? ()


Adding a sleep of 2000ms before entering the loop for send_dtmf inband permits for the test to pass the dtmf send/detection stage.
So it seems this is an issue with pjmedia_conf with too many ports (maybe we should create a pjmedia_conf per call or per AudioEndpoint).

But when processing the call termination, it crashed when processing on_state_changed (termination):

  partial_match({                                                                                                                                                                      
    event: 'call_ended',                                                                                                                                                               
    call_id: 194                                                                                                                                                                       
  }),                                                                                                                                                                                  
  partial_match({                                                                          
    event: 'call_ended',                                                                                                                                                               
    call_id: 195                                                                                                                                                                       
  }),                                                                                      
  partial_match({                                                                                                                                                                      
    event: 'call_ended',                                                                                                                                                               
    call_id: 196                                                                                                                                                                       
  }),                                                                                                                                                                                  
  partial_match({                                                                                                                                                                      
    event: 'call_ended',                                                                                                                                                               
    call_id: 197                                                                                                                                                                       
  }),                                                                                      
  partial_match({                                                                          
    event: 'call_ended',
    call_id: 198                      
  }),                                                                                      
  partial_match({
    event: 'call_ended',                    
    call_id: 199                 
  })
]
on_state_changed

e->type=5
inv->state=6
call will terminate call=edf722b8

close_media call_id=64
close_media_endpoint edf72710
close_media_transport 55ce668
audio_endpoint_remove_port
node: ../src/pj/os_core_unix.c:1436: pj_mutex_lock: Assertion `mutex' failed.

Thread 11 "clock" received signal SIGABRT, Aborted.
[Switching to Thread 0x7fffee7fc640 (LWP 254100)]
__pthread_kill_implementation (no_tid=0, signo=6, threadid=140737194739264) at ./nptl/pthread_kill.c:44
44      ./nptl/pthread_kill.c: No such file or directory.
(gdb) bt
#0  __pthread_kill_implementation (no_tid=0, signo=6, threadid=140737194739264) at ./nptl/pthread_kill.c:44
#1  __pthread_kill_internal (signo=6, threadid=140737194739264) at ./nptl/pthread_kill.c:78
#2  __GI___pthread_kill (threadid=140737194739264, signo=signo@entry=6) at ./nptl/pthread_kill.c:89
#3  0x00007ffff7842476 in __GI_raise (sig=sig@entry=6) at ../sysdeps/posix/raise.c:26
#4  0x00007ffff78287f3 in __GI_abort () at ./stdlib/abort.c:79
#5  0x00007ffff782871b in __assert_fail_base (fmt=0x7ffff79dd130 "%s%s%s:%u: %s%sAssertion `%s' failed.\n%n", assertion=0x7ffff4f8de09 "mutex", 
    file=0x7ffff4fb4af3 "../src/pj/os_core_unix.c", line=1436, function=<optimized out>) at ./assert/assert.c:92
#6  0x00007ffff7839e96 in __GI___assert_fail (assertion=assertion@entry=0x7ffff4f8de09 "mutex", file=file@entry=0x7ffff4fb4af3 "../src/pj/os_core_unix.c", line=line@entry=1436, 
    function=function@entry=0x7ffff4fb4ef0 <__PRETTY_FUNCTION__.9> "pj_mutex_lock") at ./assert/assert.c:101
#7  0x00007ffff4f47e5d in pj_mutex_lock (mutex=<optimized out>) at ../src/pj/os_core_unix.c:1436
#8  0x00007ffff4f482d6 in pj_atomic_get (atomic_var=0x7fffd8003cb8) at ../src/pj/os_core_unix.c:1096
#9  0x00007ffff4f4e5c4 in grp_lock_acquire (p=0x7fffd8003c28) at ../src/pj/lock.c:293
#10 0x00007ffff4ee2ca2 in send_rtcp (stream=stream@entry=0x7ffff7baa4f8, with_sdes=1, with_bye=with_bye@entry=0, with_fb=with_fb@entry=0, with_xr=0) at ../src/pjmedia/stream.c:1094
#11 0x00007ffff4ee2fa4 in check_tx_rtcp (timestamp=263360, stream=0x7ffff7baa4f8) at ../src/pjmedia/stream.c:1257
#12 check_tx_rtcp (stream=0x7ffff7baa4f8, timestamp=263360) at ../src/pjmedia/stream.c:1227
#13 0x00007ffff4ee46b2 in put_frame_imp (frame=frame@entry=0x7fffee7fbc60, port=<optimized out>, port=<optimized out>) at ../src/pjmedia/stream.c:1541
#14 0x00007ffff4ee4d14 in put_frame (port=0x7ffff7baad08, frame=0x7fffee7fbc60) at ../src/pjmedia/stream.c:1712
#15 0x00007ffff4ed3315 in write_port (frm_type=<synthetic pointer>, timestamp=0x7fffee7fbce8, cport=0x5495878, conf=0x7ffff7affb30) at ../src/pjmedia/conference.c:1926
#16 get_frame (this_port=<optimized out>, frame=0x7fffee7fbcd0) at ../src/pjmedia/conference.c:2221
#17 0x00007ffff4ed8d83 in clock_callback (ts=0x7ffff4cb2f80, user_data=0x7ffff4cb2e88) at ../src/pjmedia/master_port.c:193
#18 0x00007ffff4ef431b in clock_thread (arg=0x7ffff4cb2f60) at ../src/pjmedia/clock_thread.c:391
#19 0x00007ffff4f48198 in thread_main (param=0x7ffff4cb6078) at ../src/pj/os_core_unix.c:701
#20 0x00007ffff7894ac3 in start_thread (arg=<optimized out>) at ./nptl/pthread_create.c:442
#21 0x00007ffff7926850 in clone3 () at ../sysdeps/unix/sysv/linux/x86_64/clone3.S:81
(gdb) 

So it seems we are not removing some port from the conf upon call termination.

UPDATE: after the final changes for #60 I could not reproduce these problems anymore.

-------------------------
2024/03/09 takeshi:

Regarding implementation of start_speech_recog, I experimented testing with pocketsphinx in the command line but results were not good:

First I compiled pocketsphinx and a sample app:

git clone https://github.com/cmusphinx/pocketsphinx # got commit 7be89aae3e76568e02e4f3d41cf1adcb7654430c the latest at that time
cmake .
make
cd examples
gcc -o simple simple.c -I ../include/ -L ../ -lpocketsphinx -lm

Obs: i built pocketsphinx from source because the one installed by apt in ubuntu 22.04 is too old and the examples don't compil with it.


Then using flite tool in the command-line with voice kal (8000) converted to 16000 works:

takeshi@takeshi-desktop:examples$ flite "how are you" -voice kal -o c.wav
takeshi@takeshi-desktop:examples$ sox c.wav -r 16000 cc.wav
takeshi@takeshi-desktop:examples$ POCKETSPHINX_PATH=/home/takeshi/src/git/pocketsphinx/model/ ./simple cc.wav
how are you

However the audio generated by
  start_speech_synth(call_id, {"voice": "kal", "text": "how are you"}) // kal is 8000 and mapped to kal16 in flite_port.
and recorded by the other side of the call doesn't work (we get "are you" instead of "how are you"):

takeshi@takeshi-desktop:examples$ sox oc.wav -r 16000 oc2.wav
takeshi@takeshi-desktop:examples$ POCKETSPHINX_PATH=/home/takeshi/src/git/pocketsphinx/model/ ./simple oc2.wav
are you

Increasing the volume doesn't solve this:

takeshi@takeshi-desktop:examples$ sox -v 2.0 oc.wav -r 16000 oc2.wav
takeshi@takeshi-desktop:examples$ POCKETSPHINX_PATH=/home/takeshi/src/git/pocketsphinx/model/ ./simple oc2.wav
are you

So it seems there is audio degradation in the processing (perceptible by comparing the wav files by ear).
We need to solve this otherwise, we will not be able to test flite talking with pocketsphinx (well it might work if we stream using sampling rate 16000)
Audio probably degrades when flite_port use the 16000 voices and call cst_wave_resample to resample them to 8000 to permit to go via pcm.

So I tried to use pjmedia resample_port to change flite 16000 to 8000. But with both high-quality (default) and PJMEDIA_RESAMPLE_USE_LINEAR the pocketsphinx transcription quality was not good.

Obs: this is not that bad. We might still use pocketsphinx by recording messages to wav files and playing the file instead of generating the audio using flite.
Maybe with good enunciation we can have adequate results.

However, if kal (which uses sampling rate of 8000 works with flite command line tool, then we should be able to generate at that rate and have similar results.
So let's investigate this further.

Obs: I tested with speex/16000 and with it results were reasonably good.


-------------------------
2024/03/10 takeshi:

Research to implement pocketsphinx_port.c.

This worked:

takeshi@takeshi-desktop:examples$ cat stream_from_file.c 
/*

Convert c.wav (8000hz) to raw file (16000hz):
 sox c.wav -t raw -e signed -r 16000 c16000.raw pad 0 10

Play raw file:
  play -t raw -e signed -r 16000 -b 16 -c 1 c16000.raw
  
Compile app:
  gcc -o stream_from_file stream_from_file.c -I ../include/ -L ../ -lpocketsphinx -lm -lportaudi

Execute app:

takeshi@takeshi-desktop:examples$ POCKETSPHINX_PATH=/home/takeshi/src/git/pocketsphinx/model/ ./stream_from_file c16000.raw 16000 
sampling_rate=16000 frame_size=480
Speech start at 0.24
Speech end at 1.08
are you
takeshi@takeshi-desktop:examples$ 

As we can see. Instead of 'how are you' we get just 'are you'.

But testing with flite output we confirmed our implementation is OK as we also get just 'are you':

takeshi@takeshi-desktop:examples$ flite 'how are you?' -voice kal -o a.wav

takeshi@takeshi-desktop:examples$ soxi a.wav

Input File     : 'a.wav'
Channels       : 1
Sample Rate    : 8000
Precision      : 16-bit
Duration       : 00:00:01.00 = 7971 samples ~ 74.7281 CDDA sectors
File Size      : 16.0k
Bit Rate       : 128k
Sample Encoding: 16-bit Signed Integer PCM

takeshi@takeshi-desktop:examples$ sox a.wav -t raw -e signed -r 16000 a16000.raw pad 0 10
takeshi@takeshi-desktop:examples$ POCKETSPHINX_PATH=/home/takeshi/src/git/pocketsphinx/model/ ./stream_from_file a16000.raw 16000
sampling_rate=16000 frame_size=480
Speech start at 0.24
Speech end at 1.08
are you
takeshi@takeshi-desktop:examples$

Then if we use kal16 (16000hz), it is OK:

takeshi@takeshi-desktop:examples$ flite 'how are you?' -voice kal16 -o b.wav
takeshi@takeshi-desktop:examples$ soxi b.wav

Input File     : 'b.wav'
Channels       : 1
Sample Rate    : 16000
Precision      : 16-bit
Duration       : 00:00:01.00 = 15942 samples ~ 74.7281 CDDA sectors
File Size      : 31.9k
Bit Rate       : 256k
Sample Encoding: 16-bit Signed Integer PCM

takeshi@takeshi-desktop:examples$ sox b.wav -t raw -e signed -r 16000 b16000.raw pad 0 10
takeshi@takeshi-desktop:examples$ POCKETSPHINX_PATH=/home/takeshi/src/git/pocketsphinx/model/ ./stream_from_file b16000.raw 16000
sampling_rate=16000 frame_size=480
Speech start at 0.24
Speech end at 1.08
how are you
takeshi@takeshi-desktop:examples$

*/


#include <pocketsphinx.h>

int
main(int argc, char *argv[])
{
    short *frame;
    ps_decoder_t *decoder;
    ps_config_t *config;
    ps_endpointer_t *ep;
    size_t frame_size;

    (void)argc; (void)argv;

    char *filename = argv[1];
    int sampling_rate = atoi(argv[2]);

    config = ps_config_init(NULL);
    ps_default_search_args(config);

    if ((decoder = ps_init(config)) == NULL)
        E_FATAL("PocketSphinx decoder init failed\n");
    if ((ep = ps_endpointer_init(0, 0.0, 0, sampling_rate, 0)) == NULL)
        E_FATAL("PocketSphinx endpointer init failed\n");
    frame_size = ps_endpointer_frame_size(ep);
    if ((frame = malloc(frame_size * sizeof(frame[0]))) == NULL)
        E_FATAL_SYSTEM("Failed to allocate frame");

    printf("sampling_rate=%i frame_size=%lu\n", sampling_rate, frame_size);
   
    FILE *rawfd = fopen(filename, "rb");
    if (rawfd == NULL) {
        E_FATAL_SYSTEM("Failed to open raw audio file %s for reading.\n", filename);
    }

    while (!feof(rawfd)) {
    	size_t x = fread(frame, sizeof(short), frame_size, rawfd);

        const int16 *speech;
        int prev_in_speech = ps_endpointer_in_speech(ep);
        speech = ps_endpointer_process(ep, frame);
        if (speech != NULL) {
            const char *hyp;
            if (!prev_in_speech) {
                fprintf(stdout, "Speech start at %.2f\n",
                        ps_endpointer_speech_start(ep));
		fflush(stdout); /* For broken MSYS2 terminal */
                ps_start_utt(decoder);
            }
            if (ps_process_raw(decoder, speech, frame_size, FALSE, FALSE) < 0)
                E_FATAL("ps_process_raw() failed\n");
            if ((hyp = ps_get_hyp(decoder, NULL)) != NULL) {
                //fprintf(stdout, "PARTIAL RESULT: %s\n", hyp);
		fflush(stdout);
	    }
            if (!ps_endpointer_in_speech(ep)) {
                fprintf(stdout, "Speech end at %.2f\n",
                        ps_endpointer_speech_end(ep));
		fflush(stdout);
                ps_end_utt(decoder);
                if ((hyp = ps_get_hyp(decoder, NULL)) != NULL) {
		    printf("%s\n", hyp);
		    fflush(stdout);
		}
            }
        }
    }
    free(frame);
    ps_endpointer_free(ep);
    ps_free(decoder);
    ps_config_free(config);
        
    return 0;
}
takeshi@takeshi-desktop:examples$ 


The frame_size requested by pocketsphinx was 480.
I tried forcing 160, and with this, nothing was detected.
Then I tried forcing 320 and with this it worked (got 'are you' for kall and 'how are you' for kal16)
So in the pocketsphinx_port.c we will need to buffer data to the desired frame_size before feeeding it to ps.

-------------------------
2024/03/16 takeshi:

Regarding start_speech_recog, I tested debian 11 apt package but it is old and will not compile with code written for latest pocketsphinx version:

$ apt install libpocketsphinx-dev

$ gcc -o stream_from_file stream_from_file.c -I /usr/include/pocketsphinx/ -lpocketsphinx -lm
stream_from_file.c: In function ‘main’:
stream_from_file.c:78:5: error: unknown type name ‘ps_config_t’
   78 |     ps_config_t *config;
      |     ^~~~~~~~~~~
stream_from_file.c:79:5: error: unknown type name ‘ps_endpointer_t’
   79 |     ps_endpointer_t *ep;
      |     ^~~~~~~~~~~~~~~

$ cat /etc/issue
Debian GNU/Linux 11 \n \l

$ dpkg -l |grep pocketsphinx
ii  libpocketsphinx-dev:amd64        0.8+5prealpha+1-13                    amd64        Speech recognition tool - front-end library development files
ii  libpocketsphinx3:amd64           0.8+5prealpha+1-13                    amd64        Speech recognition tool - front-end library
ii  pocketsphinx-en-us               0.8+5prealpha+1-13                    all          Speech recognition tool - US English language model

So we will need to build pocketsphinx from source and link statically to it.

Regarding the pocketsphinx model we will distribute the one that comes with it.
We will store it within our module and index.js will set the path to it when start_speech_recog is specified without passing it.

-------------------------
2024/06/08 takeshi:

Here is result of tests executed inside a docker container started by doing:
  takeshi@takeshi-desktop:~$ docker run -it debian:bookworm-slim /bin/bash

After installation of sip-lab I switched to node_modules/sip-lab and runtests resulted in:
```

Success. All tests passed

Successful tests:
  - samples/100_calls.js: duration=0.00 seconds
  - samples/16_audio_streams.js: duration=0.00 seconds
  - samples/183_session_progress.js: duration=0.00 seconds
  - samples/delayed_media.js: duration=0.00 seconds
  - samples/four_audio_streams_two_refused.js: duration=0.00 seconds
  - samples/g729.js: duration=0.00 seconds
  - samples/mrcp_and_audio.js: duration=0.00 seconds
  - samples/mrcp_and_audio.simplified_media.js: duration=0.00 seconds
  - samples/multiple_audio_streams.js: duration=0.00 seconds
  - samples/options.js: duration=0.00 seconds
  - samples/pcma.js: duration=0.00 seconds
  - samples/play_wav_and_speech_recog.bad_transcript.pcmu8000.js: duration=0.00 seconds
  - samples/refer.js: duration=0.00 seconds
  - samples/refuse_telephone_event.js: duration=0.00 seconds
  - samples/register_no_expires.js: duration=0.00 seconds
  - samples/register_subscribe.js: duration=0.00 seconds
  - samples/reinvite_and_dtmf.js: duration=0.00 seconds
  - samples/reinvite_audio_audio.js: duration=0.00 seconds
  - samples/reinvite_with_hold_unhold.js: duration=0.00 seconds
  - samples/rtp_and_srtp.js: duration=0.00 seconds
  - samples/rtp_and_srtp.rtp_refused.js: duration=0.00 seconds
  - samples/send_and_receive_fax.js: duration=0.00 seconds
  - samples/simple.js: duration=0.00 seconds
  - samples/sip_cancel.js: duration=0.00 seconds
  - samples/speech_synth_and_recog.speex16000.js: duration=0.00 seconds
  - samples/srtp.js: duration=0.00 seconds
  - samples/start_play_wav_with_end_of_file_event.js: duration=0.00 seconds
  - samples/start_play_wav_with_no_loop.js: duration=0.00 seconds
  - samples/tcp.js: duration=0.00 seconds
  - samples/text_to_speech.js: duration=0.00 seconds
  - samples/tls.js: duration=0.00 seconds
  - samples/two_audio_streams.js: duration=0.00 seconds
  - samples/two_audio_streams.port_zero.js: duration=0.00 seconds

Everything OK

root@636c5c5b0748:~/tmp/t2/node_modules/sip-lab# node -v
v19.9.0
```