SIP Trunk Register/Call Issues

[ma-Hel]

Hi, first of all thank you for your great work on this project!

I am trying to use this library to make an automated call to a SIP Trunk under certain conditions and I am having trouble with the telephone system at the customers site:
I have used MicroSIP to verify the registration and the call to the SIP Trunk both work. However, when I use my custom application, I get a "404 not found" response from the telephone system. I captured the invites from sipSorcery (first) and microsip (after the new registration).

(Sorry, I cant copy files or text from the machine I am working from..)

Here is an invite created with sipSorcery (response: 404 not found):

And here is an invite created by microSIP (working):

The difference that I can make out is:

1. The "Contact" field, at the MicroSIP invite its 5060@ip, where as with the bad invite its just the ip.

2. The Authorization field is missing in the sipsorcery invite.. But I would expect a different error if the request were not authorized.

I do not have a lot of experience with SIP, so maybe something else is wrong or there is an obvious mistake somewhere.
But I cant figure out how to match the differences that I mentioned above and I dont know if they even cause the issue I am having..

What i am currently doing in code are basically the following steps:

private const string SERVER_IP = "10.135.134.132";
private const int SERVER_EXPIRY = 500;
private const string PROXY_IP = "10.135.134.132"; 
private const int PROXY_PORT = 5060; // outbound proxy port
private const string TRUNK_DOMAIN = "testdomain.com"; // no leading '@'
private const string CALLER_USER = "5060";
private const string CALLER_PW = "SIPCalling";

static SIPTransport _sipTransport = new SIPTransport();
// Use default options to set up a SIP channel.
SIPUserAgent userAgent;
SIPRegistrationUserAgent regUserAgent;


WindowsAudioEndPoint winAudio = new SIPSorceryMedia.Windows.WindowsAudioEndPoint(new SIPSorcery.Media.AudioEncoder());

_sipTransport = new SIPTransport();
_sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(System.Net.IPAddress.Any, 0)));
_sipTransport.CustomiseRequestHeader = (local, remote, req) =>
{
    if (req.Method == SIPMethodsEnum.INVITE)
    {
        req.Header.Contact = new List<SIPContactHeader> { SIPContactHeader.GetDefaultSIPContactHeader(SIPSchemesEnum.sip) };
    }
    return req.Header;

};

var outboundProxy = new SIPEndPoint(SIPProtocolsEnum.udp, System.Net.IPAddress.Parse(PROXY_IP), PROXY_PORT);

userAgent = new SIPUserAgent(_sipTransport, outboundProxy);

SIPURI AOR = new SIPURI("5060", TRUNK_DOMAIN, "");
SIPURI contact = new SIPURI("5060", "10.135.134.122", "");

regUserAgent = new SIPRegistrationUserAgent(_sipTransport, outboundProxy, AOR, CALLER_USER, CALLER_PW, "5060", TRUNK_DOMAIN, contact, SERVER_EXPIRY, [""]);
regUserAgent.RegistrationSuccessful += registerSuccessful;
regUserAgent.RegistrationFailed += registerFailed;
regUserAgent.RegistrationRemoved += registerRemoved;
regUserAgent.Start();

VoIPMediaSession voipMediaSession = new SIPSorcery.Media.VoIPMediaSession(winAudio.ToMediaEndPoints());

// setup for the media session
voipMediaSession.AcceptRtpFromAny = true;
voipMediaSession.OnAudioFormatsNegotiated += (formats) => voipMediaSession.AudioExtrasSource.SetAudioSourceFormat(formats.First());
voipMediaSession.AudioExtrasSource.AudioSamplePeriodMilliseconds = 30;

bool callOk = await userAgent.Call(destination, CALLER_USER, CALLER_PW, _voipMediaSession);

Any ideas on what could cause this behaviour / what I should do differently?

[sipsorcery]

The SIP URIs in the two posted screenshots are different. One is using 2869 and one is uing 2456. That would be the first thing I'd check.

Other than that you shouldn't need to customise the contact header in the customise method. Is there a reason you're doing that?

[ma-Hel]

Hi, thanks for you reply.
I forgot to mention that: I tested the connection with MicroSIP and sipsorcery to multiple different numbers, but I unfortunately did not document a matching pair. So i dont think that the URIs are the issue.

You are right, I copied that from somewhere out of desperation and never removed it. Adding custom headers just to use the default contact header does not make too much sense I guess..
And now I get the 5060@ip in the "contact" field, just like I do with the MicroSip Invite packet.

I will only be able to access the system at the end of next week, so right now I cant check if that solves my problem.
I ll post an update then.

[ma-Hel]

I had the chance to try with the telephone system, but I still get a "404 Not found" error. I now understand it makes sense that there is no "auth"-header in the first invite, since it is only sent after an invite was challenged, which is not happening here.

I recorded an invite sent with SipSorcery and one sent with MicroSIP:

SIPSorcery

Via: SIP/2.0/UDP 10.135.15.103:62105;branch=z9hG4bK5d82532f7e2f4111ace2da311cd8768d;rport
To: sip:2456@abc.com
From: sip:5060@abc.com;tag=TORACLPFHK
Call-ID: 5754ad1cc1ca4a51bd2e121fa71b9ef8
CSeq: 1 INVITE
Contact: sip:5060@10.135.15.103:62105
Max-Forwards: 70
User-Agent: sipsorcery_v8.0.23.0
Supported: replaces, norefersub, 100rel
Content-Length: 344

MicroSIP

Via: SIP/2.0/UDP 10.135.15.103:52216;rport;branch=z9hG4bKPjbcd320fe08a248deadff6f0227e067e5
Max-Forwards: 70
From: sip:5060@abc.com;tag=9c6d22d1e115467dae3abf5ab0964781
To: sip:2456@abc.com
Contact: sip:5060@10.135.15.103:52216;ob
Call-ID: aa709d5857b34620ae999f3db6b19877
CSeq: 8595 INVITE
Route: sip:10.135.134.132;lr
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: MicroSIP/3.22.3
Content-Type: application/sdp
Content-Length: 343

The Response to the MicroSIP invite is "401 Unauthorized", as expected, but the invite sent from my SIPsorcery client gets a "404 Not Found" response.
I was able to get a correct response with my test-setup (freeSwitch).

Do you have an idea what could be causing this issue?
Is any additional information needed to make the issue clearer?

Any help would be greatly appreciated.

[sipsorcery]

My suspciion would be the extra routing inform MicroSIP has is allowing INVITEs from it to work.

Route: sip:10.135.134.132;lr

Have you configured anything in your MicroSIP set up with the 10.135.134.132 IP address?

[ma-Hel]

Yes, I added that address as both SIP-Server and SIP-Proxy. I tried adding the route to the CallDescriptor, but the 404 error remained. However now that I think about it, I believe I may have added it with a port as well, while the MicroSIP invite doesnt have that..

[ma-Hel]

I added the "Route" entry and I noticed there was a difference between the register and invite "contact" headers, which I fixed. And now the invite goes through and the phone actually rings!
However, I do not get an audio connection.
This is the Invite that is sent:

INVITE sip:2456@abc.com SIP/2.0
Via: SIP/2.0/UDP 10.135.15.103:63432;branch=z9hG4bK4c1c1e9ccd0f4dd8a612b7ca03848625;rport
To: sip:2456@abc.com
From: sip:5060@abc.com;tag=BRDXNXFTJH
Call-ID: e23bf92c2eb343f3b2d4ef29544e028c
CSeq: 2 INVITE
Contact: sip:5060@10.135.15.103:63432
Max-Forwards: 70
Route: sip:10.135.134.132;lr
User-Agent: sipsorcery_v8.0.23.0
Authorization: Digest username="5060",realm="5060",nonce="10938cc18412cc4e75107e7acdc0346745e77f5bC16445637A7D",uri="sip:2456@abc.com",response="2eb6d88bb37b19615532edae907b5caf",algorithm=MD5
Supported: replaces, norefersub, 100rel
Content-Length: 345

v=0
o=- 2116870197 0 IN IP4 127.0.0.1
s=sipsorcery
c=IN IP4 10.135.15.103
t=0 0
m=audio 58174 RTP/AVP 0 8 9 18 101
a=mid:0
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=ssrc:1842173830 cname:207b143e-bfc4-46eb-9cde-3f5ed45b131b

For comparison, this is the MicroSIP Invite:

INVITE sip:2456@abc.com SIP/2.0
Via: SIP/2.0/UDP 10.135.15.103:60622;rport;branch=z9hG4bKPj57fa975c03f0450a93a5801388227cd2
Max-Forwards: 70
From: sip:5060@abc.com;tag=7cfd1f8af4a6432daf401af41416aa6f
To: sip:2456@abc.com
Contact: sip:5060@10.135.15.103:60622;ob
Call-ID: 4f3dc8fd996b47ff8f43df492f3c9fd8
CSeq: 30826 INVITE
Route: sip:10.135.134.132;lr
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: MicroSIP/3.22.3
Authorization: Digest username="5060", realm="5060", nonce="d181d0afa5e33fbd9e305e2cf0e4dc2590e746a8C16445D366E8", uri="sip:2456@abc.com", response="96d9a305a500f65199eaade48e3a7018", algorithm=MD5
Content-Type: application/sdp
Content-Length: 343

v=0
o=- 3978601380 3978601380 IN IP4 10.135.15.103
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4000 RTP/AVP 8 0 101
c=IN IP4 10.135.15.103
b=TIAS:64000
a=rtcp:4001 IN IP4 10.135.15.103
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:1405502687 cname:6c5561634b5d0db5

I believe the reason is in the owner row: o = ... IN IP4 127.0.0.1, a local address makes no sense here right?
If I understand the SDP Protocol Memo correctly, no private IP should be used if the message leaves the scope(?), but I saw this in other issues here as well, where it seemingly caused no problems...
I tried to set the ip in the transport layer like so:

            _sipTransport = new SIPTransport();
            var ip = System.Net.IPAddress.Parse(LOCAL_IP);
            SIPUDPChannel _sipChannel = new SIPUDPChannel(new IPEndPoint(ip, 0));
            _sipTransport.AddSIPChannel(_sipChannel);
            _sipTransport.PreferIPv6NameResolution = false;

but it did not make a difference for me,
and Copilot told me to set the RtpSession LocalEndPoint, but kept pointing me towards imaginary structures in the library.

Maybe it could also be a firewall issue, since I am working on a PC that is not in the same network as the telephone system and I rely on the customers IT to setup the ports for this..