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

[Kristopher Lalletti (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-18094?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=201497#comment-201497 ] 

Kristopher Lalletti commented on ASTERISK-18094:
------------------------------------------------

Out of curiosity, I just tried latest Polycom Firmwrae (4.0.3.7562) with ilbc 13.33kb being the only codec available, and still no luck on Asterisk 1.8.19.1

It seems like Asterisk really doesn't want to get involved in buffering different calls legs of different frame rates (which would probably mean having to carry a buffer of 60ms/120ms in memory to transcode 30ms frames into 20ms frames since I would assume that smallest frame size has to be a multiple of the larger one)


                
> 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
>
> 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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