[asterisk-bugs] [JIRA] (ASTERISK-18094) iLBC (30ms packet) to G711 (20ms) ULAW transcoding sounds "robotic"

[Rob Gagnon (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-18094?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=218903#comment-218903 ] 

Rob Gagnon commented on ASTERISK-18094:
---------------------------------------

Our company's need for this to work went away, so I have not had the time to work on it more.  The reason is that iLBC 20ms is just not the popular.  We only had one client wanting to use it, and they just changed all of their stuff to 30ms due to interoperability issues they were having with almost any carrier they connected with.  For asterisk, the buffer matching overhead would almost be a waste of time because all the other codecs are 30ms.  The effort needed to have buffering work right to convert 30ms to 20ms and vice-versa just seems to outweigh the benefits.

The pjSIP library seems to support both frame sizes, so maybe newer asterisk that uses that would work for you?

> 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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