[asterisk-bugs] [JIRA] (ASTERISK-26745) Asymmetric codecs when asymmetric_rtp_codec=no

[Jesse Ross (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-26745?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Jesse Ross updated ASTERISK-26745:
----------------------------------

    Description: 
Calling from a Grandstream HT802 (opus,ulaw) to a trunk (ulaw), causes one way audio (no audio heard on HT802)

Wireshark shows HT802 send opus and receives ulaw packets. I am able to play these received ulaw packets and hear audio. My assumption is that this ATA does not support asymmetric codecs.

I pulled the 13 and 14 branches from the git repo and had the same issue.

In my case, it was not solved by this:
https://issues.asterisk.org/jira/browse/ASTERISK-26603

Attached are 

debug.txt (verbose=10, debug=10, rtp and pjsip debugging on)
endpoint1.txt (HT802 endpoint calling from)
endpoint2.txt (sip trunk endpoint)
ht802.txt (snippet of syslog from GS ATA)

I originally made this post: https://community.asterisk.org/t/problems-with-opus-grandstream-ht802-directmedia-native-rtp/69459/5

My dialplan is quite simple so I didn't include it here. Dialed extension pass to our ARI app for processing. We are currently using the Asterisk 13 method of creating a channel and bridging (no separate create then dial).

If I disable native_rtp bridging (using "bridge technology suspend native_rtp") I do get audio in both directions. Asterisk creates a simple_bridge in this case.

If I calling using this ATA to an extension set up to just play music on hold, I can hear audio. Asterisk first sends ulaw then correctly identifies that it received a opus packet and switches to match.

  was:
Calling from a Grandstream HT802 (opus,ulaw) to a trunk (ulaw), causes one way audio (no audio heard on HT802)

Wireshark shows HT802 send opus and receives ulaw packets. I am able to play these received ulaw packets and hear audio. My assumption is that this ATA does not support asymmetric codecs.

I pulled the 13 and 14 branches from the git repo and had the same issue.

In my case, it was not solved by this:
https://issues.asterisk.org/jira/browse/ASTERISK-26603

Attached are 

debug.txt (verbose=10, debug=10, rtp and pjsip debugging on)
endpoint1.txt (HT802 endpoint calling from)
endpoint2.txt (sip trunk endpoint)
ht802.txt (snippet of syslog from GS ATA)

I originally made this post: https://community.asterisk.org/t/problems-with-opus-grandstream-ht802-directmedia-native-rtp/69459/5


My dialplan is quite simple so I didn't include it here. Dialed extension pass to our ARI app for processing. We are currently using the Asterisk 13 method of creating a channel and bridging (no separate create then dial).

If I disable native_rtp bridging (using "bridge technology suspend native_rtp") I do get audio in both directions. Asterisk creates a simple_bridge in this case.


> Asymmetric codecs when asymmetric_rtp_codec=no
> ----------------------------------------------
>
>                 Key: ASTERISK-26745
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26745
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>    Affects Versions: 13.13.1, 14.2.1, GIT
>            Reporter: Jesse Ross
>         Attachments: debug.txt, endpoint1.txt, endpoint2.txt, ht802.txt
>
>
> Calling from a Grandstream HT802 (opus,ulaw) to a trunk (ulaw), causes one way audio (no audio heard on HT802)
> Wireshark shows HT802 send opus and receives ulaw packets. I am able to play these received ulaw packets and hear audio. My assumption is that this ATA does not support asymmetric codecs.
> I pulled the 13 and 14 branches from the git repo and had the same issue.
> In my case, it was not solved by this:
> https://issues.asterisk.org/jira/browse/ASTERISK-26603
> Attached are 
> debug.txt (verbose=10, debug=10, rtp and pjsip debugging on)
> endpoint1.txt (HT802 endpoint calling from)
> endpoint2.txt (sip trunk endpoint)
> ht802.txt (snippet of syslog from GS ATA)
> I originally made this post: https://community.asterisk.org/t/problems-with-opus-grandstream-ht802-directmedia-native-rtp/69459/5
> My dialplan is quite simple so I didn't include it here. Dialed extension pass to our ARI app for processing. We are currently using the Asterisk 13 method of creating a channel and bridging (no separate create then dial).
> If I disable native_rtp bridging (using "bridge technology suspend native_rtp") I do get audio in both directions. Asterisk creates a simple_bridge in this case.
> If I calling using this ATA to an extension set up to just play music on hold, I can hear audio. Asterisk first sends ulaw then correctly identifies that it received a opus packet and switches to match.



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