[asterisk-bugs] [Asterisk 0014021]: RTP playout does not match ptime

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=14021 
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Reported By:                Skavin
Assigned To:                kpfleming
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Project:                    Asterisk
Issue ID:                   14021
Category:                   Core/RTP
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     assigned
Asterisk Version:           1.4.21.2 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2008-12-04 16:02 CST
Last Modified:              2009-11-06 03:53 CST
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Summary:                    RTP playout does not match ptime
Description: 
when a sip client invites with a alaw and ptime of 30. Asterisk sends RTP
at intervals of 20 and 40 ms as captured by tcpdump on the asterisk
server.
this is causing 20ms jitter on these connections.

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---------------------------------------------------------------------- 
 (0113270) pasandev (reporter) - 2009-11-06 03:53
 https://issues.asterisk.org/view.php?id=14021#c113270 
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I tested autoframing with asterisk 1.4.21.2

and works for me with some sip phone. netcomm v90.

When the phone specify the rtp packetization in SDP asterisk negotiate to
that frame size

Below is the SIP Packets I observed during my test.




<--- SIP read from 192.168.101.118:5060 --->
INVITE sip:2025 at 192.168.101.182:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.101.118:5060;rport;branch=z9hG4bK1794878507
From: "2026" <sip:2026 at 192.168.101.182;user=phone>;tag=5581114
To: <sip:2025 at 192.168.101.182>
Call-ID: 675324517 at 192.168.101.118:5060
CSeq: 4 INVITE
Contact: <sip:2026 at 192.168.101.118:5060>
Max-Forwards: 30
User-Agent: NETCOMM
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REFER, REGISTER,
MESSAGE
Content-Type: application/sdp
Content-Length:   223

v=0
o=NETCOMM 1257480847 1257480847 IN IP4 192.168.101.118
s=NETCOMM
c=IN IP4 192.168.101.118
t=0 0
m=audio 10000 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:100


===========================================================================================================
Response From asterisk
===========================================================================================================


<--- Reliably Transmitting (no NAT) to 192.168.101.118:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.101.118:5060;branch=z9hG4bK1653030959;received=192.168.101.118;rport=5060
From: "2026" <sip:2026 at 192.168.101.182;user=phone>;tag=5581114
To: <sip:2025 at 192.168.101.182>;tag=as27cf1d2d
Call-ID: 675324517 at 192.168.101.118:5060
CSeq: 5 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:2025 at 192.168.101.182>
Content-Type: application/sdp
Content-Length: 268

v=0
o=root 2849 2849 IN IP4 192.168.101.182
s=session
c=IN IP4 192.168.101.182
t=0 0
m=audio 18180 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:100
a=sendrecv


This means autoframing works if client specify the ptime in itz SDP


I try this test with a Mitel 5224 But Didn't work, coz Mitel doesn't
change the ptime in SDP when it configured with different RTP frame sizes 

Issue History 
Date Modified    Username       Field                    Change               
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2009-11-06 03:53 pasandev       Note Added: 0113270                          
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