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Summary:ASTERISK-24705: No sound when using WebRTC in some calls
Reporter:Juan P. Daza P. (tcpip4000)Labels:
Date Opened:2015-01-20 09:19:53.000-0600Date Closed:2015-02-24 09:02:31.000-0600
Priority:MajorRegression?
Status:Closed/CompleteComponents:Channels/chan_sip/General Channels/chan_sip/WebSocket
Versions:13.1.0 Frequency of
Occurrence		Constant
Related
Issues:
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 Attachments:( 0) http.conf
( 1) log-call-no-audio.txt
( 2) log-call-ok.txt
( 3) rtp.conf
( 4) sip.conf
Description: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.
Comments:By: Juan P. Daza P. (tcpip4000) 2015-01-20 10:00:14.957-0600

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...)

By: Rusty Newton (rnewton) 2015-01-30 14:03:08.064-0600

We need another set of logs that also includes the "DEBUG" logger channels.

You should follow the instructions here: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information

That'll also have you gather and attach the actual log file rather than copying from the CLI.

Thanks!
By: Juan P. Daza P. (tcpip4000) 2015-02-09 09:46:16.117-0600

Thank you for your response, we were using an old version of SIPML5 and after update we have sound between the parties but the DTMF tones using RFC-2833 can't be heard.   So no DTMF arrives to destination.  

I'll create a new set of logs with your recomendation.
By: Rusty Newton (rnewton) 2015-02-24 09:04:01.139-0600

{quote}
Thank you for your response, we were using an old version of SIPML5 and after update we have sound between the parties but the DTMF tones using RFC-2833 can't be heard. So no DTMF arrives to destination.
{quote}

Alright. That sounds like a new/different issue.

{quote}
I'll create a new set of logs with your recomendation.
{quote}

I've closed out this issue as cannot reproduce. Please create a new JIRA issue for your DTMF issue and attach your logs there.