[asterisk-bugs] [JIRA] (ASTERISK-18094) iLBC (30ms packet) to G711 (20ms) ULAW transcoding sounds "robotic"
Rob Gagnon (JIRA)
noreply at issues.asterisk.org
Fri Feb 8 11:03:59 CST 2013
[ https://issues.asterisk.org/jira/browse/ASTERISK-18094?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=202747#comment-202747 ]
Rob Gagnon edited comment on ASTERISK-18094 at 2/8/13 11:03 AM:
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We have seen this same problem under Asterisk 10.12.1 for iLBC
Asterisk is only able at this time to use the 30ms / 13.33Kbps version of the iLBC codec. Adding ":20" to the codec in your config file won't do anything since there is no code behind it to support it.
I'm working on a complete patch that will allow both 20ms and 30ms modes of iLBC in one codec file. For now, I've uploaded a proof-of-concept patch that simply changes the iLBC codec to 20ms only (no 30ms mode). See attachment "asterisk-10.12.1-ilbc-20ms.patch"
Feel free to try it out. In our tests, we were able to hear clearly and did see the proper "mode=20" in the SIP SDP. Our current testing still has no audio in one direction due to a faulty softphone sending 50-byte frames even though it negotiates mode=20 properly. In case you were interested, mode 20 is supposed to be 38-byte frames, so with this patch, 50-byte iLBC frames are dropped and hence: no audio.
We've been working on this actively for only about a day, so any feedback as far as any other softphones working better would be helpful.
was (Author: rgagnon):
We have seen this same problem unser Asterisk 10.12.1 for iLBC
Asterisk is only able at this time to use the 30ms / 13.33Kbps version of the iLBC codec. Adding ":20" to the codec in your config file won't do anything since there is no code behind it to support it.
I'm working on a complete patch that will allow both 20ms and 30ms modes of iLBC in one codec file. For now, I've uploaded a proof-of-concept patch that simply changes the iLBC codec to 20ms only (no 30ms mode). See attachment "asterisk-10.12.1-ilbc-20ms.patch"
Feel free to try it out. In our tests, we were able to hear clearly and did see the proper "mode=20" in the SIP SDP. Our current testing still has no audio in one direction due to a faulty softphone sending 50-byte frames even though it negotiates mode=20 properly. In case you were interested, mode 20 is supposed to be 38-byte frames, so with this patch, 50-byte iLBC frames are dropped and hence: no audio.
We've been working on this actively for only about a day, so any feedback as far as any other softphones working better would be helpful.
> iLBC (30ms packet) to G711 (20ms) ULAW transcoding sounds "robotic"
> -------------------------------------------------------------------
>
> Key: ASTERISK-18094
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-18094
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Components: Channels/chan_sip/CodecHandling
> Affects Versions: 1.8.4
> Environment: 2.6.34.8-68.fc13.i686.PAE #1 SMP Thu Feb 17 14:54:10 UTC 2011 i686 i686 i386 GNU/Linux
> Running on VMware ESX 4.0 with asterisk compiled for timerfd and dedicated CPU shares from the hypervisor.
> Reporter: Kristopher Lalletti
> Severity: Minor
> Attachments: asterisk10.12.1-ilbc-20ms.patch
>
>
> Here's the transcoding flow:
> (Polycom560 v3.3.1)==SIP/iLBC(30ms)===>(Asterisk 1.8.4.2)===SIP/G711(20ms)===>(Class-5 telco switch)==>MyLandLine
> This problem seems to be consistent since the fork-out of the iLBC source-code from the Asterisk SRC tree. The result is a "robotic" symptom, with consistently lost fragments of sound.
> When I change the flow to the following:
> (Polycom560 v3.3.1)==SIP/iLBC(30ms)===>(Asterisk 1.8.4.2)===>MeetMeBridge(provided by DAHDI/pseudo)
> (MyLandLine)===>(Class-5 telco switch)===>SIP/G711(20ms)===>(Asterisk 1.8.4.2)==>MeetMeBridge(provided by DAHDI/pseudo)
> End-to-end, the sound is fine in both directions (note: meetme.conf has audiobuffers=0 defined to ensure minimal buffering).
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