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@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@5g4f1tqaruji.invalid] has been answered

2013-06-28 11:52:01.271125 [DEBUG] switch_channel.c:3595 (sofia/internal/sip:9ku10lcl@5g4f1tqaruji.invalid) Callstate Change RINGING -> ACTIVE

2013-06-28 11:52:01.291128 [DEBUG] switch_core_codec.c:244 sofia/internal/1003@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@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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