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

[Aaron Meriwether (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-18094?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=231569#comment-231569 ] 

Aaron Meriwether commented on ASTERISK-18094:
---------------------------------------------

If you set Zoiper to use SIP rather than IAX2 to connect to Asterisk (while leaving the media codecs otherwise the same), does the issue still occur?  If not, then it might simply be a matter of porting some of the chan_sip fixes over to chan_iax2 as well.

> 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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