[asterisk-bugs] [JIRA] (ASTERISK-25197) asterisk 13.4.0 fails to transcode calls to/from g729 with TCE400P

[dant (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-25197?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=226917#comment-226917 ] 

dant commented on ASTERISK-25197:
---------------------------------

Seeing the same behaviour with Asterisk 13.3.0 going alaw<->ulaw

Call starts with one end, the source, using ulaw, the other end, the destination, using alaw, as the call comes up, bridge looks to set up native_rtp, checks the read/write codecs, finds both sides have ulaw and gets the native_rtp bridge running.
As RTP starts flowing, each packet from the destination, using alaw results in the following being logged:
[Jul 16 21:01:40] DEBUG[16971][C-00002340] res_rtp_asterisk.c: Unsupported payload type received
No Got  RTP packet debug messages are logged.

The first packet going out to the alaw host has the following:
[Jul 16 21:01:40] DEBUG[16972][C-00002340] res_rtp_asterisk.c: Ooh, format changed from none to alaw
followed by Send RTP packet debug messages.

One way audio is ulaw to alaw, no audio from alaw to ulaw.

core show channel on the outgoing (alaw) leg shows:
  NativeFormats: (alaw)
    WriteFormat: ulaw
     ReadFormat: ulaw
 WriteTranscode: Yes (ulaw at 8000)->(alaw at 8000)
  ReadTranscode: Yes (alaw at 8000)->(ulaw at 8000)

and on the incoming (ulaw) leg:
  NativeFormats: (ulaw)
    WriteFormat: ulaw
     ReadFormat: ulaw
 WriteTranscode: No
  ReadTranscode: No

I was able to work around it by adding a VOLUME function call in dialplan to attach an audiohook to prevent the bridge being eligible for native_rtp. e.g.
same => n,Set(VOLUME(RX)=1)

Looking at res_rtp_asterisk.c I can't see how the packet could be triggering the message "Unsupported payload type" in bridge_p2p_rtp_write without a "Got  RTP packet" debug message.


> asterisk 13.4.0 fails to transcode calls to/from g729 with TCE400P
> ------------------------------------------------------------------
>
>                 Key: ASTERISK-25197
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-25197
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>    Affects Versions: 13.4.0
>         Environment: SLES 11 SP3 (3.0.101-0.47.52-default x86_64)
>            Reporter: Tom Parker
>            Assignee: Tom Parker
>         Attachments: aruast.pcap, asterisk-config.tar, core show translation.txt, dahdi.txt, myDebugLog.txt
>
>
> With the TCE400P installed a call placed from a Polycom SIP phone using ulaw to a SIP trunk using g729 succeeds but audio is only one way.  The caller does not hear any audio but the callee can.  
> This also happens between two phones on the same asterisk pbx if one phone is forced to use ulaw and the other g729.  In this scenario the caller does not receive any audio from the callee.
> Downgrading asterisk to 11.13 has fixed the problem for me with no change at all in my configuration or sip setup for my telephones.



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