audio nack questions

[bryand...@gmail.com]

The audio nack feature described here mentions TabCapture:

https://code.google.com/p/webrtc/issues/detail?id=1601

Is it safe to use audio nack with regular webcam video?

Is there any reason not to use it along with video nack?

Thanks

[Niklas Enbom]

It's untested for non-streaming mode (higher delay). The risk is that you will send extra packets that won't be used since they arrive too late.

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[bryand...@gmail.com]

Is there a way to switch between streaming and non-streaming modes when not using tab capture ?  

My app is more about presentation than conversation, so a little more delay with higher quality would be preferable in most cases.

[Niklas Enbom]

There's no way to do that in Chrome.

[ravindra pai]

Hi All,

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

I am using chrome Version 30.0.1599.66 with  Linux 3.2.0-29-generic #46-Ubuntu SMP.

1) I did getUserMedia.

2) I have added a=rtcp-fb:100 nack line for audio in SDP.

"

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

"

3) I did setLocalDesciption on resulting SDP i.e pc.setLocalDescription(sessionDescription);

4) I dont get onIceCandidate callback.

Without adding a=rtcp-fb:nack line it works fine.

Please suggest me, what I am doing wrong?

Thanks in advance,

Ravindra M 

[Suhas Nandakumar]

Just noted , a=rtcp-fb point to Payload Type 100 which is not listed in the m=audio line but instead is listed in the m=video line ...

May be you wanna check that first

./S

[ravindra pai]

Thanks Suhas. a=rtcp-fb:103 worked but a=rtcp-fb:* doesnt work.

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[Vikas]

Hi,

I can recreate the problem with wild card payload types, filed a bug here.

/Vikas