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Summary:ASTERISK-16052: [patch] [regression] No T.140 text for outbound SIP calls
Reporter:peterj (peterj)Labels:
Date Opened:2010-05-04 18:33:10Date Closed:2011-06-07 14:05:21
Priority:MinorRegression?Yes
Status:Closed/CompleteComponents:Channels/chan_sip/CodecHandling
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) bug.diff
( 1) bug1.diff
Description:Sip T.140 text codec negotiation looks okay, no rtp text is setup anyway.

Im providing a patch that checks if textsupport=yes is set in sip.conf, and then enables text. This patch uses same logic as for video.

If textsupport flag isnt set the problem will remain. That is, the codec negotiation will look like text is setup, but asterisk will not have an rtp object set up to receieve text.

Im really tired when im writing this, hope it makes sense anyway

Thanks








****** ADDITIONAL INFORMATION ******

This a log from unpatched asterisk v 1.6.2.6

U 192.168.0.108:5060 -> 192.168.0.190:5060
INVITE sip:test@192.168.0.190 SIP/2.0.
Via: SIP/2.0/UDP 192.168.0.108:5060;branch=z9hG4bK5b6c91b8;rport.
Max-Forwards: 70.
From: "asterisk" <sip:asterisk@192.168.0.108>;tag=as2ad9948b.
To: <sip:test@192.168.0.190>.
Contact: <sip:asterisk@192.168.0.108>.
Call-ID: 63ca0bc4775204aa61cd6c132265320a@192.168.0.108.
CSeq: 102 INVITE.
User-Agent: Asterisk PBX 1.6.2.6.
Date: Tue, 04 May 2010 23:14:15 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
Supported: replaces, timer.
Content-Type: application/sdp.
Content-Length: 327.
.
v=0.
o=root 1427622690 1427622690 IN IP4 192.168.0.108.
s=Asterisk PBX 1.6.2.6.
c=IN IP4 192.168.0.108.
t=0 0.
m=audio 12554 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.
m=text 18334 RTP/AVP 106.
a=rtpmap:106 T140/1000.
a=sendrecv.

#
U 192.168.0.190:5060 -> 192.168.0.108:5060
SIP/2.0 100 Trying.
From: "asterisk"<sip:asterisk@192.168.0.108>;tag=as2ad9948b.
To: <sip:test@192.168.0.190>.
Call-ID: 63ca0bc4775204aa61cd6c132265320a@192.168.0.108.
CSeq: 102 INVITE.
Via: SIP/2.0/UDP 192.168.0.108:5060;rport=5060;branch=z9hG4bK5b6c91b8.
Supported: 100rel,replaces.
User-Agent: eConf4.2.71.
Accept: application/sdp,audio/telephone-event,application/media_control+xml,application/dtmf-relay,message/sipfrag,text/html,text/plain.
Contact: <sip:test@192.168.0.190>.
Content-Length: 0.
.

#
U 192.168.0.190:5060 -> 192.168.0.108:5060
SIP/2.0 180 Ringing.
From: "asterisk"<sip:asterisk@192.168.0.108>;tag=as2ad9948b.
To: <sip:test@192.168.0.190>;tag=2b6cef8-be00a8c0-13c4-40030-642e-e6d1686-642e.
Call-ID: 63ca0bc4775204aa61cd6c132265320a@192.168.0.108.
CSeq: 102 INVITE.
Via: SIP/2.0/UDP 192.168.0.108:5060;rport=5060;branch=z9hG4bK5b6c91b8.
Supported: 100rel,replaces.
User-Agent: eConf4.2.71.
Accept: application/sdp,audio/telephone-event,application/media_control+xml,application/dtmf-relay,message/sipfrag,text/html,text/plain.
Contact: <sip:test@192.168.0.190>.
Content-Length: 0.
.

#
U 192.168.0.190:5060 -> 192.168.0.108:5060
SIP/2.0 200 OK.
From: "asterisk"<sip:asterisk@192.168.0.108>;tag=as2ad9948b.
To: <sip:test@192.168.0.190>;tag=2b6cef8-be00a8c0-13c4-40030-642e-e6d1686-642e.
Call-ID: 63ca0bc4775204aa61cd6c132265320a@192.168.0.108.
CSeq: 102 INVITE.
Via: SIP/2.0/UDP 192.168.0.108:5060;rport=5060;branch=z9hG4bK5b6c91b8.
Supported: 100rel,replaces.
User-Agent: eConf4.2.71.
Accept: application/sdp,audio/telephone-event,application/media_control+xml,application/dtmf-relay,message/sipfrag,text/html,text/plain.
Contact: <sip:test@192.168.0.190>.
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,REFER,PRACK,INFO,MESSAGE,SUBSCRIBE,NOTIFY,UPDATE.
Content-Type: application/sdp.
Content-Length: 242.
.
v=0.
o=anonymous 1273014866 1273014864 IN IP4 192.168.0.190.
s=-.
i=eConf4.2.71.
c=IN IP4 192.168.0.190.
b=AS:2048.
t=0 0.
m=audio 6000 RTP/AVP 0.
a=rtpmap:0 PCMU/8000.
a=sendrecv.
m=text 6008 RTP/AVP 106.
a=rtpmap:106 T140/1000.
a=sendrecv.

#
U 192.168.0.108:5060 -> 192.168.0.190:5060
ACK sip:test@192.168.0.190 SIP/2.0.
Via: SIP/2.0/UDP 192.168.0.108:5060;branch=z9hG4bK121ad969;rport.
Max-Forwards: 70.
From: "asterisk" <sip:asterisk@192.168.0.108>;tag=as2ad9948b.
To: <sip:test@192.168.0.190>;tag=2b6cef8-be00a8c0-13c4-40030-642e-e6d1686-642e.
Contact: <sip:asterisk@192.168.0.108>.
Call-ID: 63ca0bc4775204aa61cd6c132265320a@192.168.0.108.
CSeq: 102 ACK.
User-Agent: Asterisk PBX 1.6.2.6.
Content-Length: 0.


Comments:By: Leif Madsen (lmadsen) 2010-05-07 11:13:14

Thanks for the submission! You might want to ping the asterisk-dev mailing list about this as well for testers and to make sure this is the appropriate approach. Thanks!
By: Leif Madsen (lmadsen) 2010-05-11 10:30:23

Just pinging you about this issue as this seems right up your alley :)
By: David Vossel (dvossel) 2010-05-28 15:30:05

I just want to make sure this isn't a configuration issue first.  Have you referred to the doc/realtimetext.txt help document in the Asterisk source tree to verify you configured it correctly?
By: Leif Madsen (lmadsen) 2010-06-08 09:47:17

Pinging the reporter -- any feedback on this?
By: Paul Belanger (pabelanger) 2010-06-16 09:21:42

Suspended due to lack of activity. Please request a bug marshal in #asterisk-bugs on the IRC network irc.freenode.net to reopen the issue should you have the additional information requested.

Further information can be found at http://www.asterisk.org/developers/bug-guidelines