[asterisk-bugs] [JIRA] (ASTERISK-24705) No sound when using WebRTC in some calls

[Rusty Newton (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-24705?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Rusty Newton updated ASTERISK-24705:
------------------------------------

    Component/s:     (was: . I did not set the category correctly.)
                 Channels/chan_sip/WebSocket
                 Channels/chan_sip/General

> No sound when using WebRTC in some calls
> ----------------------------------------
>
>                 Key: ASTERISK-24705
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24705
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/General, Channels/chan_sip/WebSocket
>    Affects Versions: 13.1.0
>         Environment: SERVER
> NAME=openSUSE
> VERSION="13.1 (Bottle)"
> VERSION_ID="13.1"
> PRETTY_NAME="openSUSE 13.1 (Bottle) (x86_64)"
> kernel = 3.11.10-21-default
> processor =  Intel Xeon E312xx (Sandy Bridge)
> asterisk = 13.1.0
> gcc = 4.8.1
> CLIENT
> Windows 7 64 bits
> Chrome = 39.0.2171.99 (64-bit)
> SIPML5
>            Reporter: Juan P. Daza P.
>         Attachments: http.conf, log-call-no-audio.txt, log-call-ok.txt, rtp.conf, sip.conf
>
>
> When using SIPML5 phone in chrome to make a call it works as expected when the number is a landline call.
> When using the same webphone calling a cellphone number there is no audio.
> The difference I found in the logs is a line that says something like:
>      Probation passed - setting RTP source address to
> When that line shows up the RTP traffic can be seen in the log and the audio is transmitted, otherwise no audio is transmitted but the dtmf tones can be hear if buttons pressed.



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