You could use proxy_mode to avoid messing with the SDP at all. set the option inbound-proxy-media to true in the profile and FS will not get involved in the codecs On Thu, Nov 6, 2008 at 3:57 AM, shehzad p <pmhshz@gmail.com> wrote:
In the following trace, aaa.aaa.aaa.aaa = originator bbb.bbb.bbb.bbb = freeswitch ccc.ccc.ccc.ccc = terminator
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# U 2008/11/06 03:26:22.208150 aaa.aaa.aaa.aaa:5060 -> bbb.bbb.bbb.bbb:5060 INVITE sip:XXXXXXXXXXX@bbb.bbb.bbb.bbb SIP/2.0. Via: SIP/2.0/UDP aaa.aaa.aaa.aaa:5060;branch=z9hG4bK50b7762d;rport. From: "1598" <sip:1598@aaa.aaa.aaa.aaa>;tag=as0178868b. To: <sip:XXXXXXXXXXX@bbb.bbb.bbb.bbb>. Contact: <sip:1598@aaa.aaa.aaa.aaa>. Call-ID: 651149c41338f2801f8059825e048eaa@aaa.aaa.aaa.aaa. CSeq: 102 INVITE. User-Agent: Asterisk PBX. Max-Forwards: 70. Date: Thu, 06 Nov 2008 09:41:55 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Supported: replaces. Content-Type: application/sdp. Content-Length: 261. . v=0. o=root 2055 2055 IN IP4 aaa.aaa.aaa.aaa. s=session. c=IN IP4 aaa.aaa.aaa.aaa. t=0 0. m=audio 13262 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv.
# U 2008/11/06 03:26:22.208352 bbb.bbb.bbb.bbb:5060 -> aaa.aaa.aaa.aaa:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP
aaa.aaa.aaa.aaa:5060;branch=z9hG4bK50b7762d;rport=5060;received=aaa.aaa.aaa.aaa. From: "1598" <sip:1598@aaa.aaa.aaa.aaa>;tag=as0178868b. To: <sip:XXXXXXXXXXX@bbb.bbb.bbb.bbb>. Call-ID: 651149c41338f2801f8059825e048eaa@aaa.aaa.aaa.aaa. CSeq: 102 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.1-exported. Content-Length: 0. .
# U 2008/11/06 03:26:22.233253 bbb.bbb.bbb.bbb:5080 -> ccc.ccc.ccc.ccc:5060 INVITE sip:5651XXXXXXXXXXX@ccc.ccc.ccc.ccc SIP/2.0. Via: SIP/2.0/UDP bbb.bbb.bbb.bbb:5080;rport;branch=z9hG4bK10QUmt8DjFDmF. Max-Forwards: 69. From: "1598" <sip:1598@bbb.bbb.bbb.bbb>;tag=4eyQv6ZZZNBBK. To: <sip:5651XXXXXXXXXXX@ccc.ccc.ccc.ccc>. Call-ID: 6f5f1baf-267f-122c-1b8c-003048911494. CSeq: 106861655 INVITE. Contact: <sip:mod_sofia@bbb.bbb.bbb.bbb:5080>. User-Agent: FreeSWITCH-mod_sofia/1.0.1-exported. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO. Supported: 100rel, timer, precondition, path, replaces. Allow-Events: talk. Min-SE: 120. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 279. Remote-Party-ID: "1598" <sip:1598@bbb.bbb.bbb.bbb>;screen=yes;privacy=off. . v=0. o=FreeSWITCH 4182297201107985931 4664015740918096257 IN IP4 bbb.bbb.bbb.bbb. s=FreeSWITCH. c=IN IP4 bbb.bbb.bbb.bbb. t=0 0. a=sendrecv. m=audio 28302 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20.
# U 2008/11/06 03:26:22.420544 ccc.ccc.ccc.ccc:5060 -> bbb.bbb.bbb.bbb:5080 SIP/2.0 100 Trying. Via: SIP/2.0/UDP bbb.bbb.bbb.bbb:5080;rport;branch=z9hG4bK10QUmt8DjFDmF. From: "1598" <sip:1598@bbb.bbb.bbb.bbb>;tag=4eyQv6ZZZNBBK. To: <sip:5651XXXXXXXXXXX@ccc.ccc.ccc.ccc>. Call-ID: 6f5f1baf-267f-122c-1b8c-003048911494. CSeq: 106861655 INVITE. Content-Length: 0. .
# U 2008/11/06 03:26:22.433039 ccc.ccc.ccc.ccc:5060 -> bbb.bbb.bbb.bbb:5080 SIP/2.0 488 Not Acceptable Here. Supported: 100rel. Via: SIP/2.0/UDP bbb.bbb.bbb.bbb:5080;rport;branch=z9hG4bK10QUmt8DjFDmF. To: <sip:5651XXXXXXXXXXX@ccc.ccc.ccc.ccc>;tag=3434953230-169789. From: "1598" <sip:1598@bbb.bbb.bbb.bbb>;tag=4eyQv6ZZZNBBK. Remote-Party-Id: "1598" <sip:1598@bbb.bbb.bbb.bbb>;screen=yes;privacy=off. Call-ID: 6f5f1baf-267f-122c-1b8c-003048911494. CSeq: 106861655 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK. Contact: <sip:5651XXXXXXXXXXX@ccc.ccc.ccc.ccc:5060>. Call-Info: <sip:ccc.ccc.ccc.ccc>;method="NOTIFY;Event=telephone-event;Duration=1000". Content-Length: 0. .
# U 2008/11/06 03:26:22.433137 bbb.bbb.bbb.bbb:5080 -> ccc.ccc.ccc.ccc:5060 ACK sip:5651XXXXXXXXXXX@ccc.ccc.ccc.ccc SIP/2.0. Via: SIP/2.0/UDP bbb.bbb.bbb.bbb:5080;rport;branch=z9hG4bK10QUmt8DjFDmF. Max-Forwards: 69. From: "1598" <sip:1598@bbb.bbb.bbb.bbb>;tag=4eyQv6ZZZNBBK. To: <sip:5651XXXXXXXXXXX@ccc.ccc.ccc.ccc>;tag=3434953230-169789. Call-ID: 6f5f1baf-267f-122c-1b8c-003048911494. CSeq: 106861655 ACK. Content-Length: 0. .
# U 2008/11/06 03:26:22.454386 bbb.bbb.bbb.bbb:5060 -> aaa.aaa.aaa.aaa:5060 SIP/2.0 488 Not Acceptable Here. Via: SIP/2.0/UDP
aaa.aaa.aaa.aaa:5060;branch=z9hG4bK50b7762d;rport=5060;received=aaa.aaa.aaa.aaa. From: "1598" <sip:1598@aaa.aaa.aaa.aaa>;tag=as0178868b. To: <sip:XXXXXXXXXXX@bbb.bbb.bbb.bbb>;tag=vmZD3BgNZQ22c. Call-ID: 651149c41338f2801f8059825e048eaa@aaa.aaa.aaa.aaa. CSeq: 102 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.1-exported. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. Supported: 100rel, timer, precondition, path, replaces. Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary. Reason: Q.850;cause=16;text="NORMAL_CLEARING". Content-Length: 0. .
# U 2008/11/06 03:26:22.813849 aaa.aaa.aaa.aaa:5060 -> bbb.bbb.bbb.bbb:5060 ACK sip:XXXXXXXXXXX@bbb.bbb.bbb.bbb SIP/2.0. Via: SIP/2.0/UDP aaa.aaa.aaa.aaa:5060;branch=z9hG4bK50b7762d;rport. From: "1598" <sip:1598@aaa.aaa.aaa.aaa>;tag=as0178868b. To: <sip:XXXXXXXXXXX@bbb.bbb.bbb.bbb>;tag=vmZD3BgNZQ22c. Contact: <sip:1598@aaa.aaa.aaa.aaa>. Call-ID: 651149c41338f2801f8059825e048eaa@aaa.aaa.aaa.aaa. CSeq: 102 ACK. User-Agent: Asterisk PBX. Max-Forwards: 70. Content-Length: 0.
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Check the above trace. "annexb=no" comes from originator to freeswitch, but while sending to terminator removes that.
Please Refer the trace and help me for the problem.
Brian West-3 wrote:
I would need to see a sip trace of this taking place. If you're using the passthru codec we do pass the fmtp options thru when we receive them.
/b
On Nov 5, 2008, at 8:26 AM, shehzad p wrote:
I have to route the inbound calls of G729A codec. Calls comes to my freeswitch with codec G729A (As "annexb=no" is set)
But when i route calls to termination gateway, calls are dropped (because of "annexb=no " is not set)
Why "annexb=no" is removed while i route the calls? How can I set "annexb=no'? (I am using javascript for routing the calls)
Does following SDP variables can help me in solving above problem? How to use those variables? http://wiki.freeswitch.org/wiki/Channel_Variables#SDP_Manipulation
Warm thanks in advance... MSP -- View this message in context:
http://www.nabble.com/Problem-in-Routing-G729A-Calls-tp20342694p20342694.htm...
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