[Freeswitch-users] Problem in Routing G729A Calls

Thu Nov 6 06:19:03 PST 2008

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.
> >>
> >>
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> --
> 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.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20081106/6dd816c0/attachment-0002.html 