[Freeswitch-users] Problem in Routing G729A Calls

Thu Nov 6 07:30:26 PST 2008

What if I need to stay in media path in some cases.
Is there a way to prevent this issue?

Anthony Minessale-2 wrote:
> 
> 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 at gmail.com> wrote:
> 
>>
>> In the following trace,
>> aaa.aaa.aaa.aaa = originator
>> bbb.bbb.bbb.bbb = freeswitch
>> ccc.ccc.ccc.ccc = terminator
>>
>> ====================================================================
>>
>> #
>> U 2008/11/06 03:26:22.208150 aaa.aaa.aaa.aaa:5060 -> bbb.bbb.bbb.bbb:5060
>> INVITE sip:XXXXXXXXXXX at bbb.bbb.bbb.bbb SIP/2.0.
>> Via: SIP/2.0/UDP aaa.aaa.aaa.aaa:5060;branch=z9hG4bK50b7762d;rport.
>> From: "1598" <sip:1598 at aaa.aaa.aaa.aaa>;tag=as0178868b.
>> To: <sip:XXXXXXXXXXX at bbb.bbb.bbb.bbb>.
>> Contact: <sip:1598 at aaa.aaa.aaa.aaa>.
>> Call-ID: 651149c41338f2801f8059825e048eaa at 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 at aaa.aaa.aaa.aaa>;tag=as0178868b.
>> To: <sip:XXXXXXXXXXX at bbb.bbb.bbb.bbb>.
>> Call-ID: 651149c41338f2801f8059825e048eaa at 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 at 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 at bbb.bbb.bbb.bbb>;tag=4eyQv6ZZZNBBK.
>> To: <sip:5651XXXXXXXXXXX at ccc.ccc.ccc.ccc>.
>> Call-ID: 6f5f1baf-267f-122c-1b8c-003048911494.
>> CSeq: 106861655 INVITE.
>> Contact: <sip:mod_sofia at 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 at 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 at bbb.bbb.bbb.bbb>;tag=4eyQv6ZZZNBBK.
>> To: <sip:5651XXXXXXXXXXX at 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 at ccc.ccc.ccc.ccc>;tag=3434953230-169789.
>> From: "1598" <sip:1598 at bbb.bbb.bbb.bbb>;tag=4eyQv6ZZZNBBK.
>> Remote-Party-Id: "1598"
>> <sip:1598 at 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 at 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 at 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 at bbb.bbb.bbb.bbb>;tag=4eyQv6ZZZNBBK.
>> To: <sip:5651XXXXXXXXXXX at 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 at aaa.aaa.aaa.aaa>;tag=as0178868b.
>> To: <sip:XXXXXXXXXXX at bbb.bbb.bbb.bbb>;tag=vmZD3BgNZQ22c.
>> Call-ID: 651149c41338f2801f8059825e048eaa at 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 at bbb.bbb.bbb.bbb SIP/2.0.
>> Via: SIP/2.0/UDP aaa.aaa.aaa.aaa:5060;branch=z9hG4bK50b7762d;rport.
>> From: "1598" <sip:1598 at aaa.aaa.aaa.aaa>;tag=as0178868b.
>> To: <sip:XXXXXXXXXXX at bbb.bbb.bbb.bbb>;tag=vmZD3BgNZQ22c.
>> Contact: <sip:1598 at aaa.aaa.aaa.aaa>.
>> Call-ID: 651149c41338f2801f8059825e048eaa at aaa.aaa.aaa.aaa.
>> CSeq: 102 ACK.
>> User-Agent: Asterisk PBX.
>> Max-Forwards: 70.
>> Content-Length: 0.
>>
>> ====================================================================
>>
>> 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.html
>> >> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>> >>
>> >>
>> >
>> > _______________________________________________
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>> >
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>> >
>> >
>>
>> --
>> View this message in context:
>> http://www.nabble.com/Problem-in-Routing-G729A-Calls-tp20342694p20358106.html
>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>
>>
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>>
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
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