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43 changes: 28 additions & 15 deletions docs/FreeSWITCH-Explained/Modules/mod_unimrcp_6586728.mdx
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Expand Up @@ -152,23 +152,36 @@ MRCP version 2 is currently a draft standard, but is supported by some MRCP serv
</include>
```

* **client-ext-ip** \- External SIP IP address of MRCP client
* **client-ip** \- SIP IP address of MRCP client
* **client-port** \- SIP port of MRCP client (make sure it doesn't conflict with conf/sip\_profiles)
* **server-ip** \- SIP IP address of MRCP server
* **server-port** \- SIP port of MRCP server
* **force-destination** \-

* **client-ext-ip** \- External SIP IP address of MRCP client, not set by default
* **client-ip** \- SIP IP address of MRCP client, defaults to "127.0.0.1"
* **client-port** \- SIP port of MRCP client (make sure it doesn't conflict with conf/sip\_profiles), defaults to "5090"
* **server-ip** \- SIP IP address of MRCP server, defaults to "127.0.0.1"
* **server-port** \- SIP port of MRCP server, defaults to 5060.
* **server-username** \- SIP server user name, defaults to blank.
* **force-destination** \- Force destination IP address. Should be used only in case SDP contains incorrect connection address (local IP address behind NAT). Defaults to 0 (false)
* **sip-transport** \- "udp" or "tcp"
* **ua-name** \- User agent name
* **sdp-origin** \-
* **rtp-ext-ip** \- External IP address for client RTP
* **rtp-ip** \- IP address for client RTP
* **rtp-port-min** \- Start of RTP port range
* **rtp-port-max** \- End of RTP port range
* **playout-delay** \-
* **max-playout-delay** \-
* **ptime** \- ptime to negotiate with MRCP server
* **ua-name** \- User agent name, defaults to "FreeSWITCH"
* **sdp-origin** \- defaults to "FreeSWITCH"
* **sip-trace** \- Enables transport logging if set to "yes", "on", "true", "t", "enabled", "active", or "allow". Defaults to disabled.
* **rtp-ext-ip** \- External IP address for client RTP, not set by default.
* **rtp-ip** \- IP address for client RTP, defaults to "127.0.0.1"
* **rtp-port-min** \- Start of RTP port range, defaults to "4000"
* **rtp-port-max** \- End of RTP port range, defaults to "5000"
* **playout-delay** \- Jitter buffer initial playout delay in milliseconds. Defaults to 0.
* **min-playout-delay** \- Jitter buffer minimum playout delay in milliseconds. Defaults to 0.
* **max-playout-delay** \- Jitter buffer maximum playout delay in milliseconds.
* **ptime** \- ptime (audio duration per packet) in milliseconds to negotiate with MRCP server
* **codecs** \- codec negotiation preference. Server will probably support PCMU, PCMA, or L16
* **rtcp** \- Enable (1) or disable (0) RTCP. Defaults to 0, disabled.
* **rtcp-bye** \- RTCP BYE transmission policy. 0 == disabled, 1 == transmit RTCP BYE at end of sessions, 2 == transmit RTCP BYE at the end of each talkspurt (input). Defaults to 0, disabled.
* **rtcp-tx-interval** \- RTCP report transmission interval in milliseconds, defaults to 0 == disabled.
* **rtcp-rx-resolution** \- timeout in milliseconds to check for a new RTCP message, defaults to 0 == disabled.
* **jsgf-mime-type** \- MIME type to use for JSGF grammars, defaults to "application/x-jsgf"
* **gsl-mime-type** \- MIME type to use for GSL grammars, defaults to "application/x-nuance-gsl"
* **srgs-xml-mime-type** \- MIME type to use for SRGS XML grammars, defaults to "application/srgs+xml"
* **srgs-mime-type** \- MIME type to use for SRGS ABNF grammars, defaults to "application/srgs"
* **ssml-mime-type** \- MIME type to use for SSML (TTS), defaults to "application/ssml+xml"

##### Sample profiles

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