From studio.pqv9 at gmail.com Fri Mar 9 14:15:54 2018 From: studio.pqv9 at gmail.com (Penelope Vennisse) Date: Fri, 9 Mar 2018 14:15:54 +0100 Subject: [SIP Beyond VoIP] Stereo with sylk-webrtc Message-ID: Hi, we've been trying for a couple of days to get stereo working in Chrome with the sylk-webrtc browser project. Modifying the SDP and setting echoCancellation to false have not had any effect. Is there a known issue with sylk-webrtc and sylkserver, for stereo with OPUS? Appreciate any feedback. We've tried what works for other webrtc apps: 1. Force Chrome to disable audio processing: constraints.audio = { mandatory: { echoCancellation: false } }; 2. Set the SDP to prefer OPUS, add stereo and high-bitrate. We also tried removing all other codecs to see if that helped. v=0 o=- 2951467701722993374 2 IN IP4 127.0.0.1 s=- t=0 0 a=msid-semantic: WMS a=group:BUNDLE audio m=audio 9 RTP/SAVPF 111 c=IN IP4 0.0.0.0 a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10;useinbandfec=1; stereo=1; sprop-stereo=1; maxplaybackrate=524288; sprop-maxcapturerate=524288; maxaveragebitrate=524288 a=rtcp:9 IN IP4 0.0.0.0 a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=setup:active a=mid:audio a=recvonly a=ice-ufrag:Q2dY a=ice-pwd:ixK6U9znPvXZeQCOzUZSJgJ/ a=fingerprint:sha-256 9D:32:5E:A5:06:6C:01:53:C4:BE: 08:F9:6B:F2:3D:83:CD:F6:51:B9:A4:5B:F1:34:F9:13:B4:C3:60:9F:2C:F6 a=rtcp-mux The connection works fine and the audio is using the bitrate directive, but it's always mono. Advice is welcome! We've verified that stereo works in the browser with other apps, for example the appr.tc reference app using https://appr.tc/r/955394799?stereo=true&audio=echoCancellation=false (Note that Chrome 65 and up currently has a stereo regression bug so we're testing in Chrome 59 using Electron until that's resolved) -------------- next part -------------- An HTML attachment was scrubbed... URL: From tijmen at ag-projects.com Fri Mar 9 17:06:28 2018 From: tijmen at ag-projects.com (Tijmen de Mes) Date: Fri, 9 Mar 2018 17:06:28 +0100 Subject: [SIP Beyond VoIP] Stereo with sylk-webrtc In-Reply-To: References: Message-ID: <3ED73AF3-13A4-45CD-987B-2B11625799D1@ag-projects.com> Hi, I think it all depends on who/what you are calling. Does the stereo not work when having an audio between Party A and Party B both on WebRTC, or does it not work when having a Video conference? Without modifying anything, I see Firefox already sending stereo=1 when making a call (local SDP), for chrome I don?t see it right now. Best regards, Tijmen de Mes ? AG Projects > On 9 mrt. 2018, at 14:15, Penelope Vennisse wrote: > > Hi, we've been trying for a couple of days to get stereo working in Chrome with the sylk-webrtc browser project. Modifying the SDP and setting echoCancellation to false have not had any effect. Is there a known issue with sylk-webrtc and sylkserver, for stereo with OPUS? Appreciate any feedback. > > We've tried what works for other webrtc apps: > > 1. Force Chrome to disable audio processing: > > constraints.audio = { > mandatory: { > echoCancellation: false > } > }; > > > 2. Set the SDP to prefer OPUS, add stereo and high-bitrate. We also tried removing all other codecs to see if that helped. > > v=0 > o=- 2951467701722993374 2 IN IP4 127.0.0.1 > s=- > t=0 0 > a=msid-semantic: WMS > a=group:BUNDLE audio > m=audio 9 RTP/SAVPF 111 > c=IN IP4 0.0.0.0 > a=rtpmap:111 opus/48000/2 > a=fmtp:111 minptime=10;useinbandfec=1; stereo=1; sprop-stereo=1; maxplaybackrate=524288; sprop-maxcapturerate=524288; maxaveragebitrate=524288 > a=rtcp:9 IN IP4 0.0.0.0 > a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level > a=setup:active > a=mid:audio > a=recvonly > a=ice-ufrag:Q2dY > a=ice-pwd:ixK6U9znPvXZeQCOzUZSJgJ/ > a=fingerprint:sha-256 9D:32:5E:A5:06:6C:01:53:C4:BE:08:F9:6B:F2:3D:83:CD:F6:51:B9:A4:5B:F1:34:F9:13:B4:C3:60:9F:2C:F6 > a=rtcp-mux > > > The connection works fine and the audio is using the bitrate directive, but it's always mono. Advice is welcome! We've verified that stereo works in the browser with other apps, for example the appr.tc reference app using https://appr.tc/r/955394799?stereo=true&audio=echoCancellation=false > > > (Note that Chrome 65 and up currently has a stereo regression bug so we're testing in Chrome 59 using Electron until that's resolved) > > _______________________________________________ > SIPBeyondVoIP mailing list > SIPBeyondVoIP at lists.ag-projects.com > http://lists.ag-projects.com/mailman/listinfo/sipbeyondvoip -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 496 bytes Desc: Message signed with OpenPGP using GPGMail URL: From l.mayer3112 at gmail.com Fri Mar 30 04:20:01 2018 From: l.mayer3112 at gmail.com (Larry Mayer) Date: Fri, 30 Mar 2018 02:20:01 -0000 Subject: [SIP Beyond VoIP] Making outbound call through SylkServer Message-ID: Hi, I have created a custom plugin which exposes a web interface. From that interface I would like to trigger an outbound call (e.g. API /startCall?n=+1234567890). The issue is I cannot find a way to start the outbound call because I would have to create an account/session/audio stream and I don't find any easy way to populate those values to make the outbound call. Should it be possible? Thank you for you support, Larry -------------- next part -------------- An HTML attachment was scrubbed... URL: