[asterisk-bugs] [Asterisk 0010355]: RTP Stream with wrong Timestamp after 200 ok when 183 session in progress
noreply at bugs.digium.com
noreply at bugs.digium.com
Tue Mar 4 12:02:20 CST 2008
The following issue has been ASSIGNED.
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http://bugs.digium.com/view.php?id=10355
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Reported By: wdecarne
Assigned To: file
======================================================================
Project: Asterisk
Issue ID: 10355
Category: Channels/chan_sip/General
Reproducibility: always
Severity: minor
Priority: normal
Status: assigned
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:
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Date Submitted: 08-01-2007 09:43 CDT
Last Modified: 03-04-2008 12:01 CST
======================================================================
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
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----------------------------------------------------------------------
svnbot - 03-04-08 12:01
----------------------------------------------------------------------
Repository: asterisk
Revision: 105674
U branches/1.4/channels/chan_sip.c
U branches/1.4/include/asterisk/rtp.h
U branches/1.4/main/rtp.c
------------------------------------------------------------------------
r105674 | file | 2008-03-04 12:01:46 -0600 (Tue, 04 Mar 2008) | 8 lines
When a new source of audio comes in (such as music on hold) make sure the
marker bit gets set.
(closes issue http://bugs.digium.com/view.php?id=10355)
Reported by: wdecarne
Patches:
10355.diff uploaded by file (license 11)
(closes issue http://bugs.digium.com/view.php?id=11491)
Reported by: kanderson
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http://svn.digium.com/view/asterisk?view=rev&revision=105674
Issue History
Date Modified Username Field Change
======================================================================
03-04-08 12:01 svnbot Note Added: 0083331
03-04-08 12:01 svnbot Status ready for testing =>
assigned
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