[Freeswitch-users] Problem in Routing G729A Calls

Thu Nov 6 01:57:34 PST 2008

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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