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

[Juan P. Daza P. (JIRA)]

Juan P. Daza P. created ASTERISK-24705:
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             Summary: 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.


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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