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

[Juan P. Daza P. (JIRA)]

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

Juan P. Daza P. updated ASTERISK-24705:
---------------------------------------

    Attachment: log-call-no-audio.txt
                sip.conf
                http.conf
                rtp.conf
                log-call-ok.txt

log-call-ok.txt: The correct call to a landline number (3731733)
log-call-no-audio.txt: Call with no sound to a cellphone number (314792...)


> 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: . I did not set the category correctly.
>    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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