|Summary:||ASTERISK-18566: G.729 RTP Payload Size|
|Date Opened:||2011-09-16 04:36:18||Date Closed:||2011-11-21 14:07:44.000-0600|
|Versions:||126.96.36.199 188.8.131.52||Frequency of|
|Environment:||Attachments:||( 0) jitter.png|
|Description:||I am not entirely sure this is a bug, but here goes.|
I was trying to create an interconnection with an AVAYA call center and from the AVAYA side we would constantly get 20ms Jitter. While digging though the packet dumps, I've realized that while my asterisks were configured to use G.729 with 20ms payload, my carrier would use 40ms payload sizes.
What strikes me as odd is that on the media path terminating on my carrier, I would get a packet with 40ms payload, and then after 20ms an empty packet. (Thus causing the jitter of 20ms.) It looks that this is not correct but since my knowledge on the matter is limited I am not sure.
I've attached a wireshark screenshot displaying the case as I've described it.
G729 was pass-through (no transcoding). Forcing the payload size to 40 on all paths resolved the issue.
|Comments:||By: m0bius (m0bius) 2011-09-16 04:36:40.082-0500|
By: Olle Johansson (oej) 2011-09-20 15:41:11.419-0500
This is an effect of the Asterisk RTP engine. We're not really sending unless we're getting a packet, so if we're getting packets every 40th ms, we will send and you will get glitches.
Did the avaya signal any packetization in the SDP? If so, there's an option in sip conf to follow whatever they say. Please show us the SDP in the INVITE or 200 OK from avaya.
By: m0bius (m0bius) 2011-09-21 05:34:45.473-0500
Here is the SDP from the 200 OK sent back by the AVAYA.
Internet Protocol, Src: 192.168.32.10 (192.168.32.10), Dst: 192.168.31.36 (192.168.31.36)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
Status-Line: SIP/2.0 200 OK
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): - 1 2 IN IP4 192.168.32.10
Session Name (s): -
Connection Information (c): IN IP4 192.168.32.11
Bandwidth Information (b): AS:64
Time Description, active time (t): 0 0
Media Description, name and address (m): audio 2578 RTP/AVP 18 101
Media Attribute (a): rtpmap:18 G729/8000
Media Attribute (a): fmtp:18 annexb=no
Media Attribute (a): rtpmap:101 telephone-event/8000
The AVAYA was configured at 20ms payload as all of my asterisks. My originating carrier was set at 40ms, so I guess the SDP I should get should be from that side.
By: Leif Madsen (lmadsen) 2011-09-21 08:31:08.006-0500
Could you provide the SIP trace of this being setup so we can see what is going on? I don't think this is a bug -- if you set it to 20ms and the other end was at 40ms, I wouldn't expect things to place nice. It would be nice to validate if we could be doing something though to make it work better or if we're doing something wrong.
By: Leif Madsen (lmadsen) 2011-09-22 08:04:17.906-0500
Oops I guess I commented at the same time as everyone else, or my screen wasn't refreshed.
By: Leif Madsen (lmadsen) 2011-09-22 08:04:34.889-0500
Requesting additional feedback from Olle.
By: Olle Johansson (oej) 2011-09-27 10:12:30.565-0500
What do you need from me Leif? We still haven't seen the other sides SDP, which is the interesting part.
By: Leif Madsen (lmadsen) 2011-10-31 11:43:48.708-0500
Reassigned to m0bius as oej has requested additional information.
By: Leif Madsen (lmadsen) 2011-11-21 14:07:34.624-0600
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