[asterisk-bugs] [JIRA] (ASTERISK-25166) No audio when using direct media and a codec with a dynamic payload

[Thomas Sevestre (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-25166?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=246545#comment-246545 ] 

Thomas Sevestre commented on ASTERISK-25166:
--------------------------------------------

Hello, Is this going to be merged in asterisk 13 ?


> No audio when using direct media and a codec with a dynamic payload
> -------------------------------------------------------------------
>
>                 Key: ASTERISK-25166
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-25166
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/CodecHandling, Resources/res_pjsip
>    Affects Versions: 13.4.0
>            Reporter: Kevin Harwell
>         Attachments: 11-sip.log.txt, 13-res_pjsip.log.txt, 13-sip.log.txt, pjsip.conf, sip.conf
>
>
> I was able to duplicate this issue in Asterisk 11 (sip) and 13 (sip and res_pjsip).
> If two endpoints are configured to use a codec with a dynamic payload and that payload number is something different than what Asterisk expects for the given codec then when using direct media the wrong payload is sent in the SDP of the reinvites. This results in no audio being heard.
> Steps to reproduce:
> 1. Using chan_sip or res_pjsip configure two endpoints to use direct media and only allow a codec with a dynamic payload.
> 2. Make a call and either one way or no audio heard.
> Note: I used g726 as the allowed codec, but you also have to make sure the payload being sent to Asterisk from the phone is different that what Asterisk uses for the payload type (111 in this case). The cisco phone I had allowed me to configure the payload numbers for g726. I also had an Aastra phone that sent g726 out with a different payload number (115) and I believe Linphone uses a different number as well.



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