Chrome M29 WebRTC Release Notes

[Vikas Marwaha]

Hi,

Sorry had missed posting M29 release notes. Please find below WebRTC Release Notes for Chrome M29. Below is the list of major changes:

Features:-

• Issue 1171 :- Added support for TURN/TCP. Now you can connect to TURN server using TCP to allocate a UDP port.
  
• Issue 1804 :- SSLTCP (pseudo-SSL with fake handshake and unencrypted data) support for p2p socket added in Chrome.
  
• Issue 1154 :- Added detection of proxy setting in browser and establishing p2p connections through the proxies.
  
• Issue 1395 :- Support proper 44.1 Khz re-sampling. Earlier, WebRTC had represented 44.1 kHz rates with 44 kHz. This caused clock drift for the AEC, now with this fixed AEC quality should improve.
  
• Issue 1745 :- Added support for AEC stats via getStats api. This is helpful in diagnosing AEC issues.
  
• Issue 1750 :- Added support for encoding 1080p video resolution, earlier we could decode 1080p but not encode. 
  
• Issue 1869 :- Added support for SDP signaling for Audio NACK. 
  
• Issue 165931 :- WebRTC enabled by default for chrome on Android. No flag needed.
  
• Issue 160494 :- Frame Rate constraints for GetUserMedia should work on linux, earlier they were ignored.
  

Bug fixes :-

• Issue 1343 :- OnIceCandidate not called for more than 10 peer connections.
  
• Issue 1487 :- Remove step delay in allocation of candidates.
  
• Issue 1578 :- WebRTC connects to TURN servers in the order you provided them.
  
• Issue 1733 :- SSLFingerprint::GetRfc4752Fingerprint has misnamed counterpart.
  
• Issue 1789 :- Wrongly closed the audio track of a mediastream while closing another stream.
  
• Issue 1828 :- Wrong treatment of unknown a=extmap attributes.
  
• Issue 1831 :- STUN binding indications are parsed incorrectly.
  
• Issue 1839 :- Crash in addAllocatedport when using STUN/TURN with peerconnection_client example.
  
• Issue 244218 :- Screen capture broken on Mac, captures on black screen.
  
• Issue 267443 :- Screen captures in WebRTC triggers a crash.
  
• Issue 266048 :- AGC sets volume to 0.
  
• Issue 167263 :- getUserMedia infobar cannot be reinvoked once user has selected Deny.

/Vikas

[PhistucK]

Thank you very much!

☆PhistucK

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[Philipp Hancke]

• Issue 1869 :- Added support for SDP signaling for Audio NACK. 

 

I'd expect to see an a=rtcp-fb line the audio media section of the localdescription?

horay at some other features like TURN/TCP :-)

[Vikas]

Hi,

I think by default the audio NACK is not used only video NACK is being used. You can refer to this thread. This feature added support for enabling audio NACK through a=rtcp-fb line in the audio media section.

/Vikas

[ravindra pai]

Chrome Version 30.0.1599.66 with Linux 3.2.0-29-generic #46-Ubuntu SMP.

adding a=rtcp-fb:100 nack for audio fails in setLocalDescription.

Steps:

1) Do getUserMedia and add a=rtcp-fb:100 nack line for audio.

v=0

o=- 6250478506366729399 2 IN IP4 127.0.0.1

s=-

t=0 0

a=group:BUNDLE audio video

a=msid-semantic: WMS 2ppkMWRHsy98od7XJmRSutkJKjt5hR0pB2m7

m=audio 1 RTP/SAVPF 111 103 104 0 8 126

c=IN IP4 0.0.0.0

a=rtcp:1 IN IP4 0.0.0.0

a=ice-ufrag:KZUq7Bfj/5FitC70

a=ice-pwd:Gn1HcVHX5UeSawJY5CETt7sQ

a=ice-options:google-ice

a=mid:audio

a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level

a=rtcp-mux

a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:ovTJj3jBDOOFV+1xCCYLRzDjk34Liwuvkl5yQTxv

a=rtpmap:111 opus/48000/2

a=fmtp:111 minptime=10

a=rtpmap:103 ISAC/16000

a=rtpmap:104 ISAC/32000

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:126 telephone-event/8000

a=maxptime:60

a=rtcp-fb:100 nack

a=ssrc:309477331 cname:IwtwGPvJ7WGO7DjD

a=ssrc:309477331 msid:2ppkMWRHsy98od7XJmRSutkJKjt5hR0pB2m7 2ppkMWRHsy98od7XJmRSutkJKjt5hR0pB2m7a0

a=ssrc:309477331 mslabel:2ppkMWRHsy98od7XJmRSutkJKjt5hR0pB2m7

a=ssrc:309477331 label:2ppkMWRHsy98od7XJmRSutkJKjt5hR0pB2m7a0

m=video 1 RTP/SAVPF 100 116 117

c=IN IP4 0.0.0.0

a=rtcp:1 IN IP4 0.0.0.0

a=ice-ufrag:KZUq7Bfj/5FitC70

a=ice-pwd:Gn1HcVHX5UeSawJY5CETt7sQ

a=ice-options:google-ice

a=mid:video

a=extmap:2 urn:ietf:params:rtp-hdrext:toffset

a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time

a=sendrecv

a=rtcp-mux

a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:ovTJj3jBDOOFV+1xCCYLRzDjk34Liwuvkl5yQTxv

a=rtpmap:100 VP8/90000

a=rtcp-fb:100 ccm fir

a=rtcp-fb:100 nack

a=rtcp-fb:100 goog-remb

a=rtpmap:116 red/90000

a=rtpmap:117 ulpfec/90000

a=ssrc:1424719162 cname:IwtwGPvJ7WGO7DjD

a=ssrc:1424719162 msid:2ppkMWRHsy98od7XJmRSutkJKjt5hR0pB2m7 2ppkMWRHsy98od7XJmRSutkJKjt5hR0pB2m7v0

a=ssrc:1424719162 mslabel:2ppkMWRHsy98od7XJmRSutkJKjt5hR0pB2m7

a=ssrc:1424719162 label:2ppkMWRHsy98od7XJmRSutkJKjt5hR0pB2m7v0

2) Do pc.setLocalDescription(sessionDescription) above sdp.

3) onIceCandidate callback is not called.

Same callflow, without adding a=rtcp-fb:100 nack line works fine.

Please help, if there is anything wrong with above SDP.

[Philipp Hancke]

Please help, if there is anything wrong with above SDP.

payload type 100 is not defined for the audio m-line.

[ravindra pai]

Thanks Philipp.

"

m=audio 1 RTP/SAVPF 111 103 104 0 8 126

c=IN IP5 0.0.0.0

a=rtcp:1 IN IP4 0.0.0.0

a=ice-ufrag:ujB1YOlul6m6qcMk

a=ice-pwd:mhNuFkML/u+aLASD3e3RtMxh

a=ice-options:google-ice

a=mid:audio

a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level

a=sendrecv

a=rtcp-mux

a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:r6Ky4z92Bb0n/h+GXlxohidUEG4RSeO1UoyFfdYb

a=rtpmap:111 opus/48000/2

a=fmtp:111 minptime=10

a=rtpmap:103 ISAC/16000

a=rtpmap:104 ISAC/32000

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:126 telephone-event/8000

a=maxptime:60

a=rtcp-fb:* nack

"

However , a=rtcp-fb:* nack fails but a=rtcp-fb:103 nack for ISAC succeeds.