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

Rusty Newton (JIRA) noreply at issues.asterisk.org
Tue Feb 24 09:05:34 CST 2015


    [ https://issues.asterisk.org/jira/browse/ASTERISK-24705?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=225084#comment-225084 ] 

Rusty Newton commented on ASTERISK-24705:
-----------------------------------------

{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.

> 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.
>            Assignee: Rusty Newton
>         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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