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Group created to discuss WebRTC. New members are moderated. Expect delays when posting as a new member.
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How to get microphone and video data together from the same page
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I use this snippet to get microphone and video data via the getUserMedia method: navigator.webkitGetUserMedia({ audio: false, video: true}, function(xtream) { video.src = URL.createObjectURL(xtream); ... navigator.webkitGetUserMedia({ audio: true, video: false}, function(stream) { var microphone = context.createMediaStreamSourc e(stream);... more »
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session rehydration clarification?
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In JSEP ([link]) section 3.6 talks about session rehydration: " With rehydration, the current signaling state is persisted somewhere outside of the page, perhaps on the application server, or in browser local storage. The page is then reloaded, the saved signaling state... more »
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Utilizing the ice-lite feature
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I have a situation where I want to use ice-lite in an outbound (from Chrome) media session. When I get back the SDP createOffer success, I inject a "a=ice-lite" just after the "c=" connection info line. I leave everything else the same, including the other ice attributes, then set that modified SDP via setLocalDescription into peerConnection which later goes... more »
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WebRTC when only TCP port 80 and 443 are open, and all UDP blocked.
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Hello. I have a rfc5766-turn-server installed on EC2 and listening on port 443 instead of 3478, and it works well for WebRTC in Chrome. If one is on a cooporate network where only port 443 and 80 are open for TCP, and all UDP are blocked (except maybe UDP port 53 that are sometimes used by DNS), it seems to me that WebRTC is not working even when you have... more »
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m-line-index vs mid when adding ice candidates
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What is the current plan WRT to sdpMLineIndex and sdpMid in the
RTCIcecandidate constructor?
chrome is happy to figure out the index given a mid, firefox isn't as I
just found out the hard way.
draft-ivov-mmusic-trickle-ice- 01 seems to have abandoned the plan to add an
a=m-line-index.
The w3c spec seems to consider the mid as optional (if present, ...).... more »
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Releasing the webcam and closing local-video
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Hello guys, a colleague and I are trying to implement video-chat-functionality to our platform with webrtc. Most things just work fine, but at the moment we have one problem: After finishing the video-chat the webcam is still on. We could pause the video-element and remove the source of it to hide the local video but at a... more »
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Problem with handling initial INVITE without the SDP
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Hi All,
We are facing trouble in handling the initial INVITE without SDP.
It means that the offer will go in 200 OK and answer will come in ACK. However, media path is not established.
Is anyone aware of such issues?
Ready! webphone.js:585
[INFO] RTC constructor qoffeesip.js:50
[INFO] New state OFFLINE(0) qoffeesip.js:1502... more »
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some changes in apprtc.py(r3840)
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Hi there, I saw there was some updates in apprtc.py (r3840) compared to r3836 about using TURN server. Some of the updated lines are: 318* turn_url = '[link] 378* turn_url = turn_url + 'turn?' + 'username=' + user + '&key=4080218913'* 393* 'turn_url': turn_url*... more »
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