Not sure what you’re doing but webrtc.freeswitch.org is running: FreeSWITCH Version 1.5.11b+git~20140307T043256Z~f9f36993e8~64bit (git f9f3699 2014-03-07 04:32:56Z 64bit) Everything sounds fine, you should try the code from now. -- Brian West brian@freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 7, 2014, at 5:07 AM, Jayesh Nambiar <jayesh1017@gmail.com> wrote:
Hi, I've been working on Freeswitch and WebRTC applications for quite some time. I was working on Freeswitch version (git c811580 2013-09-06 00:55:50Z) and the voice quality was always good with that version. Today I upgraded to the latest master version (git f9f3699 2014-03-07 04:32:56Z 64bit) and suddenly the sound that goes out of the WebRTC client is always distorted/choppy. I also tried with few versions which got committed in February but they also had choppy sound.
The call scenario is as follows: Normal SIP Endpoint calls a WebRTC endpoint. I have enforced PCMU for the leg going towards WebRTC as I had read about some problems using OPUS codec with freeswitch here https://code.google.com/p/webrtc/issues/detail?id=2768. My outgoing dial-string looks as below: {sip_invite_domain=${context},absolute_codec_string=PCMU,media_webrtc=true}sofia/abc.com/${sip_req_uri}
Sofia parameters are as follows: <param name="sip-trace" value="off"/> <param name="log-auth-failures" value="true"/> <param name="sip-ip" value="203.XXX.123.57"/> <param name="rtp-ip" value="203.XXX.123.57"/> <param name="sip-port" value="5000"/> <param name="context" value="abc.com"/> <param name="dtmf-type" value="rfc2833"/> <param name="rfc2833-pt" value="101"/> <param name="dtmf-duration" value="2000"/> <param name="caller-id-in-from" value="true"/> <param name="caller-id-type" value="pid"/> <param name="suppress-cng" value="true"/> <param name="inbound-codec-prefs" value="OPUS,G722,PCMU,PCMA"/> <param name="outbound-codec-prefs" value="PCMU,PCMA"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="300"/> <param name="outbound-use-uuid-as-callid" value="true"/>
I generally encountered such distorted quality with WebRTC when CN was advertised in SDP going towards JS-SIP. This is the reason I added suppress-sng as true in the profile. My outgoing SDP is as follows:
v=0. o=FreeSWITCH 1394161698 1394161699 IN IP4 203.XXX.123.57. s=FreeSWITCH. c=IN IP4 203.XXX.123.57. t=0 0. a=msid-semantic: WMS lBMrtzG9kpA0bTRb2XqCeNEeEVGsBGsM. m=audio 17482 RTP/SAVPF 0 101. a=rtpmap:101 telephone-event/8000. a=fingerprint:sha-256 A2:DD:5A:FE:03:98:BB:59:A5:67:EE:D2:B1:DF:B9:E7:84:7C:D0:1D:C2:68:39:EF:60:E6:5B:48:E9:72:CB:5B. a=rtcp-mux. a=rtcp:17482 IN IP4 203.XXX.123.57. a=ssrc:2871096156 cname:744rtBqDQQuSAcTt. a=ssrc:2871096156 msid:lBMrtzG9kpA0bTRb2XqCeNEeEVGsBGsM a0. a=ssrc:2871096156 mslabel:lBMrtzG9kpA0bTRb2XqCeNEeEVGsBGsM. a=ssrc:2871096156 label:lBMrtzG9kpA0bTRb2XqCeNEeEVGsBGsMa0. a=ice-ufrag:09V4Nq9hcFADbSg9. a=ice-pwd:Maq6BHzioU0OWK7M. a=candidate:5046006301 1 udp 659136 203.XXX.123.57 17482 typ host generation 0. a=candidate:5046006301 2 udp 659136 203.XXX.123.57 17482 typ host generation 0. a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:sV6KRgjWRIajcY4QqbNBQeUxxjh90KbdtfAwdmo2. a=silenceSupp:off - - - -. a=ptime:20.
Is there anything that I can add or remove to fix this quality problem or any new channel variable related to webrtc that has been added but not documented anywhere. The sound going towards JS-SIP sounds acceptable but the sound going outside JS-SIP is distorted. Any help here will be appreciated as this quality problem doesn't allow me to move to new versions and stay updated !!
Thanks,
--- Jayesh _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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