[asterisk-bugs] [Asterisk 0010355]: RTP Stream with wrong Timestamp after 200 ok when 183 session in progress

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=10355 
====================================================================== 
Reported By:                wdecarne
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   10355
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.4.9  
SVN Branch (only for SVN checkouts, not tarball releases): N/A  
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
====================================================================== 
Date Submitted:             08-01-2007 09:43 CDT
Last Modified:              08-03-2007 10:19 CDT
====================================================================== 
Summary:                    RTP Stream with wrong Timestamp after 200 ok when
183 session in progress
Description: 
Call Flow
ISDN -> Gateway -> SIP -> Asterisk Server -> SIP -> Gateway (same) ->
ISDN

GW->Asterisk (Call 1)
INVITE sip:0xxxxxxx at 172.16.0.26 SIP/2.0
From: <sip:0yyyyyyy at 172.16.0.26>;tag=-115824375
To: <sip:0xxxxxxx at 172.16.0.26>
Content-Type: application/sdp
Content-Length: 249

v=0
o=- 1185978976 1185978977 IN IP4 192.168.1.156
s=-
c=IN IP4 192.168.1.156
t=0 0
m=audio 40028 RTP/AVP 8 0 18 4 2
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:2 G726-32/8000
a=sendrecv

Asterisk->GW (Call 1)
SIP/2.0 100 Trying

Asterisk->GW (Call 2)
INVITE sip:0xxxxxxx at 192.168.1.156 SIP/2.0
From: "0yyyyyyy" <sip:0yyyyyyy at 172.16.0.26>;tag=as12435432
To: <sip:0xxxxxxx at 192.168.1.156>
Content-Type: application/sdp
Content-Length: 227

v=0
o=root 1899 1899 IN IP4 172.16.0.26
s=session
c=IN IP4 172.16.0.26
t=0 0
m=audio 30004 RTP/AVP 0 3 8
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv


GW->Asterisk  (Call 2)
SIP/2.0 100 Trying


GW->Asterisk  (Call 2)
SIP/2.0 180 Ringing
From: "0yyyyyyy" <sip:0yyyyyyy at 172.16.0.26>;tag=as12435432
To: <sip:0xxxxxxx at 192.168.1.156>;tag=-352630980
Content-Length: 0


Asterisk->GW  (Call 1)
SIP/2.0 180 Ringing
From: <sip:0yyyyyyy at 172.16.0.26>;tag=-115824375
To: <sip:0xxxxxxx at 172.16.0.26>;tag=as30f5ada3
Content-Length: 0


Asterisk->GW  (Call 1)
SIP/2.0 183 Session Progress
From: <sip:0yyyyyyy at 172.16.0.26>;tag=-115824375
To: <sip:0xxxxxxx at 172.16.0.26>;tag=as30f5ada3
Content-Type: application/sdp
Content-Length: 204

v=0
o=root 1899 1899 IN IP4 172.16.0.26
s=session
c=IN IP4 172.16.0.26
t=0 0
m=audio 30002 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

------
now
RTP Stream for Call 1
------

GW->Asterisk  (Call 2)
SIP/2.0 200 OK
From: "0yyyyyyy" <sip:0yyyyyyy at 172.16.0.26>;tag=as12435432
To: <sip:0xxxxxxx at 192.168.1.156>;tag=-352630980
Content-Type: application/sdp
Content-Length: 148

v=0
o=- 1185978979 1185978980 IN IP4 192.168.1.156
s=-
c=IN IP4 192.168.1.156
t=0 0
m=audio 40032 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=sendrecv

Asterisk->GW  (Call 2)
ACK sip:0xxxxxxx at 192.168.1.156:5060 SIP/2.0
From: "0yyyyyyy" <sip:0yyyyyyy at 172.16.0.26>;tag=as12435432
To: <sip:0xxxxxxx at 192.168.1.156>;tag=-352630980
Contact: <sip:0yyyyyyy at 172.16.0.26>
Content-Length: 0


Asterisk->GW  (Call 1)
SIP/2.0 200 OK
From: <sip:0yyyyyyy at 172.16.0.26>;tag=-115824375
To: <sip:0xxxxxxx at 172.16.0.26>;tag=as30f5ada3
Content-Length: 204

v=0
o=root 1899 1900 IN IP4 172.16.0.26
s=session
c=IN IP4 172.16.0.26
t=0 0
m=audio 30002 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

GW->Asterisk  (Call 1)
ACK sip:0xxxxxxx at 172.16.0.26 SIP/2.0
From: <sip:0yyyyyyy at 172.16.0.26>;tag=-115824375
To: <sip:0xxxxxxx at 172.16.0.26>;tag=as30f5ada3
Content-Length: 0

------
here begin the problem

the RTP stream for call 2 from Astrisk begin with the RTP Timestamp from
Call 1
and the RTP Stream for Call 1 uses the next Sequence Number but the 
RTP Timestamp is 0
====================================================================== 

---------------------------------------------------------------------- 
 wdecarne - 08-03-07 10:19  
---------------------------------------------------------------------- 
Yes, one point is that the calculation for the jitterbuffer is incorrect.
But an other point is, that a codec from AudioCodes play no audio for a few
seconds. I think the codec is waiting for the following timestamp from the
rtp stream with the calling tone and at this time the caller can't hear the
callee for a few seconds and this after the 200 respond.
It works fine when the progressinband is disable or silence suppression in
the codecs is enable (while the marker bit is use). 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
08-03-07 10:19  wdecarne       Note Added: 0068371                          
======================================================================