[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:              08-02-2007 06:37 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 - 08-02-07 06:37  
---------------------------------------------------------------------- 
Okay, i understood the behavior with the timestamp from call 1 and call 2
and the forwarding from the rtp stream.
I read the RFC3551 about the use of the marker bit and i found this:
RFC3551:
"Applications without silence suppression MUST set the marker bit to
zero."

In this application is no silence suppression. 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
08-02-07 06:37  wdecarne       Note Added: 0068302                          
======================================================================