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Summary:ASTERISK-15725: SIP RTP audio delay
Reporter:S Harvanek (sharvanek)Labels:
Date Opened:2010-03-01 20:42:34.000-0600Date Closed:2011-06-07 14:00:55
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) audio_delay.mp3
Description:This is *not* a latency issue.

Here's a full description:
http://forums.digium.com/viewtopic.php?f=1&t=73267&start=0&sid=404b57776485f8f2e376bfaf1d394dd5

This is a very very odd issue, when a caller comes in through two asterisk boxes via SIP the IVR is heard fine, but when transferred from the IVR the caller and the person being called experience about 2-3 seconds of silence before audio begins to flow.

This has been reported numerous times across the net but no real solution, some examples are:

http://www.mail-archive.com/asterisk@uc.org/msg05938.html
http://www.trixbox.org/forums/trixbox-forums/help/3-second-delay-answering-calls

sadly there isn't much debug here, a packet capture shows RTP flowing etc the same in both circumstances, please advise.

Comments:By: Leif Madsen (lmadsen) 2010-03-03 09:34:08.000-0600

I think a recording of the delay in question may also be useful here...
By: S Harvanek (sharvanek) 2010-03-03 11:42:14.000-0600

Uploaded a copy showing the issue, the "Hello this is Josh" is our side, on a Polycom IP550, you can hear the delay in the remote side not answering then a 'hello' a few seconds in.

This to me means the phone is sending proper audio to the server and asterisk has ack'd the answer but has some weird delay in forwarding voice to `Josh's` phone.
By: Leif Madsen (lmadsen) 2010-03-03 13:30:49.000-0600

Please provide a packet capture of the call from start to finish. This would include:

* SIP trace from the Asterisk console
* Asterisk console with debug level logging
* SIP history

Also what might be useful would be a full pcap with the RTP so the call can be played back from the start which demonstrates this issue.

You should also provide the relevant parts of your sip.conf file.
By: Leif Madsen (lmadsen) 2010-03-17 10:45:13

Without the information requested, we will be unable to move this issue forward, and thus we will need to close it as suspended shortly.
By: Leif Madsen (lmadsen) 2010-03-23 12:50:23

Suspended due to lack of feedback from the reporter. If you're able to provide the information requested, then feel free to reopen the issue and attach it. Thanks!