[Freeswitch-users] FreeSWITCH adds WebRTC support to new 1.4 BETA.

[Henry Huang]

no, I have specifically set bypass media to false. And if I enable bypass
media, the call from softphone to chrome won't even ring/answer. I was able
to answer from chrome, but without audio. I will generate more detailed
debug in a bit.

Henry


On Fri, Jun 28, 2013 at 12:11 PM, Michael Jerris <mike at jerris.com> wrote:

> That sdp is not for webrtc.  Are you doing bypass media from normal sip to
> webrtc?
>
> On Jun 28, 2013, at 2:56 PM, Henry Huang <red.rain.seven at gmail.com> wrote:
>
> Hi,
>
> I am getting no audio when calling from regular sip client to
> Chrome/JSsip. FreeSWITCH version is the latest (Version 1.5.3b git a52a604
> 2013-06-28 16:05:27Z)
> And JsSIP directly from their demo site.
>
> Audio works if I am calling from browser to softphone. But no audio at all
> if calling back from softphone to browser. What kind of information would
> be good to help debug this?
>
> 2013-06-28 11:52:01.271125 [NOTICE] sofia.c:6517 Channel
> [sofia/internal/sip:9ku10lcl at 5g4f1tqaruji.invalid] has been answered
> 2013-06-28 11:52:01.271125 [DEBUG] switch_channel.c:3595
> (sofia/internal/sip:9ku10lcl at 5g4f1tqaruji.invalid) Callstate Change
> RINGING -> ACTIVE
> 2013-06-28 11:52:01.291128 [DEBUG] switch_core_codec.c:244 sofia/internal/
> 1003 at 72.1.46.122 Restore previous codec PCMU:0.
> 2013-06-28 11:52:01.291128 [DEBUG] mod_sofia.c:822 Local SDP
> sofia/internal/1003 at 72.1.46.122:
> v=0
> o=FreeSWITCH 1372428618 1372428620 IN IP4 72.1.46.122
> s=FreeSWITCH
> c=IN IP4 72.1.46.122
> t=0 0
> m=audio 16892 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
>
>
>
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