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

Birger "WIMPy" Harzenetter (JIRA) noreply at issues.asterisk.org
Tue Jul 17 21:31:21 CDT 2012


    [ https://issues.asterisk.org/jira/browse/ASTERISK-18094?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=194963#comment-194963 ] 

Birger "WIMPy" Harzenetter commented on ASTERISK-18094:
-------------------------------------------------------

Looks like I ran into the same issue, but with IAX2 and without ilbc.
I blamed zoiper for not coping with 10ms frames, but just found out that it depends on what the other end of the call is. Both sip and IAX configured for alaw:10.
sip>iax2:   ok
lcr>iax2:   robotic
lcr>sip:    ok
dahdi>iax2: robotic
dahdi>sip:  ok
file>iax2:  ok

> 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: kris2k
>            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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