Firefox 53 WebRTC and Web Audio Release Notes

[Maire Reavy]

Firefox 53 is now our Developer Edition version, and it's scheduled to go to Beta on March 6th and to Release on April 18th.  Detailed WebRTC (plus Web Audio) release notes for Firefox 53 are here.

If anyone has any questions about the release, the notes, or possible bugs in the code, feel free to reach out to me directly.

Cheers,

-Maire

--------------------------------

Noteworthy Changes:

• bug 1250356 Updated core webrtc.org code to stable branch 49
  • Required major rewrite of interfaces for for video
• bug 1331498 Updated libvpx to 1.6.1 (major update)
  • Our thanks to Johann Koenig of Google for contributing the patches to update it and improve the update and build process!
• bug 1056934 TURN/TLS is now supported
• bug 1231848 and bug 1330676 etc - MediaRecorder improvements - supports >30fps, significantly improved encoding quality
• bug 1221574 Full-duplex audio for android is landed and available; currently pref'd on by default in 54, off in 53
• bug 1308481/bug 1320101/bug 1330318 Resolved TIAS bitrate limit issues after renegotiation
• bug 1322503 Fixed RTCStatsType to be spec-compatible (missing hyphens in most enum names)

--

Maire Reavy

mre...@mozilla.com, m...@reavy.org

[Henry.C]

Hi Maire,

On FF 53 beta, the video PLI rtcp packets have their "Packet sender SSRC" a random number, instead of the SSRC specified in its "answer/offer". This cause the other peer to drop the PLI packets. Consequently, lost of a key frame may take time to recover.

I have tried to bisect the code history and found this random number was assigned at

media/webrtc/signaling/src/media-conduit/VideoConduit.cpp:1152

in the commit 2233e03ef6fc.

Is it reasonable?

Regards,

Henry Chen

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[Nils Ohlmeier]

Hi Henry,

Thanks for the report. We identified the same issue and it should fixed already https://bugzilla.mozilla.org/show_bug.cgi?id=1337810

Please let us know if this is still an issue for you.

Best

  Nils Ohlmeier

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[Tri Nguyen]

Hi Maire,

On Firefox 53, I have a small RTCDataChannel-based demo which used to work on Firefox 52 and before, the same code work on Google Chrome 58. In my effort to find what is going wrong I find that if I create the data channel like follow:

const rtcDataChannel = rtcPeerConnection.createDataChannel('comm', {

            ordered: false,

            maxPacketLifeTime: this._conf.DATACHANNEL_MAX_PACKET_LIFETIME,

            // maxRetransmit: 3,

        });

 

Then after WebRTC SDP and ICE exchange on two Firefox 53 tabs on same computer, the data channel failed to open even though I confirmed that the ICE candidate pair is sucessfully formed and tested in about:webrtc.

The strange thing is if I comment maxPacketLifeTime and un-comment maxRetransmit the DataChannel could open and exchange data normally. I could verify that the unmodified version of the code works on Firefox 52 and before and on Chrome.

I attached a snapshot of about:webrtc in case this might help shed some light in this issue.

[Nils Ohlmeier]

Hello Tri,

Thanks for reporting this. I was able to reproduce the problem locally. It looks to me like the support for ‘maxPacketLifeTime’ is broken as soon as the value is non-zero.

I opened a bug report for you: https://bugzilla.mozilla.org/show_bug.cgi?id=1365081

Best

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<aboutWebrtc.html>

[Nils Ohlmeier]

Hello Tri,

On May 15, 2017, at 15:57, Nils Ohlmeier <nohl...@mozilla.com> wrote:

Thanks for reporting this. I was able to reproduce the problem locally. It looks to me like the support for ‘maxPacketLifeTime’ is broken as soon as the value is non-zero.

I opened a bug report for you: https://bugzilla.mozilla.org/show_bug.cgi?id=1365081

In case you haven’t followed the bug report: it is fixed now in Firefox Nightly.

Best

  Nils