Imaged modified from Window in Seattle airport by Flickr user Robert Scoble (CC BY-2.0)
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Multi-party calling architectures are a common topic here at webrtcHacks, largely because group calling is widely needed but difficult to implement and understand. Most would agree Scalable Video Coding (SVC) is the most advanced, but the most complex multi-party calling architecture.
To help explain how it works we have brought in not one, but two WebRTC video architecture experts. Sergio Garcia Murillo is a long time media server developer and founder of Medooze. Most recently, and most relevant for this post, he has been working on an open source SFU that leverages VP9 and SVC (the first open source project to do this that I am aware of). In addition, frequent webrtcHacks guest author and renown video expert Gustavo Garcia Bernando joins him.
Slack is an über popular and fast growing communications tool that has a ton of integrations with various WebRTC services. Slack acquired a WebRTC company a year ago and launched its own audio conferencing service earlier this year which we analyzed here and here. Earlier this week they launched video. Does this work the same? Are there any tricks we can learn from their implementation? Long time WebRTC expert and webrtcHacks guest author Gustavo Garica takes a deeper dive into Slack’s new video conferencing feature below to see what’s going on under the hood.
Dealing with multi-party video infrastructure can be pretty daunting. The good news is the technology, products, and standards to enable economical multiparty video in WebRTC has matured quite a bit in the past few years. One of the key underlying technologies enabling some of this change is called simulcast. Simulcast has been an occasional sub-topic here at webrtcHacks in the past and it is time we gave it more dedicated attention.
To do that we asked Oscar Divorra Escoda, Tokbox’s Senior Media Scientist and Media Cloud Engineering Lead to walk us through it. Tokbox was one of the first to market with a SFU and Oscar shares some of his learnings below.
Earlier this month Fippo published a post analyzing Slack’s new WebRTC implementation. He did not have direct access or a team account to do a thorough deep dive – not to mention he is supposed to be taking some off this month. That left many with some open questions? Is there more to the TURN network? How does multi-party calling work? How exactly is Slack using the Janus gateway? Fortunately WebRTC has an awesomely active and capable community that quickly picked up the slack (pun intended).
Dear Slack,
There has been quite some buzz this week about you and WebRTC.
WebRTC… kind of. Because actually you only do stuff in Chrome and your native apps:
I’ve been there. Launching stuff only for Chrome. That was is late 2012. In 2016, you need to have a very good excuse to launch something with WebRTC and not support Firefox like this:
Maybe you had your reasons. As usual, I tried to get a dump from chrome://webrtc-internals to see what is going on. Thanks to Dag-Inge Aas for providing one. The most interesting bit is the call to setRemoteDescription:
|
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 |
type: answer, sdp: v=0 o=- 1242503183783 1242503183783 IN IP4 127.0.0.1 s=Room with no name.. t=0 0 a=group:BUNDLE audio a=msid-semantic: WMS janus m=audio 1 RTP/SAVPF 111 c=IN IP4 10.9.4.95 a=mid:audio a=recvonly a=rtcp-mux a=ice-ufrag:rKmT a=ice-pwd:AOrIm8kDxew5sVNUCQrrmJ a=ice-options:trickle a=fingerprint:sha-256 C5:5F:DA:7D:84:47:B1:BF:6B:55:16:62:48:31:3E:D3:F1:7B:25:89:92:4A:4B:4D:4D:D9:D5:AF:EA:D8:15:44 a=setup:active a=connection:new a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10; useinbandfec=1; usedtx=1 a=candidate:1 1 udp 2013266431 10.9.4.95 12000 typ host a=candidate:2 1 udp 2013266431 172.31.0.190 12000 typ host |
I would like to note that you reply to Chrome’s offer of UDP/TLS/RTP/SAVPF with a profile of RTP/SAVPF. While that is still tolerated by browsers, it is improper.
Your a=msid-semantic line looks very interesting. “WMS janus”. Sounds familiar, this is meetecho’s janus gateway (see Lorenzo’s post on gateways here). Which by the way works fine with Firefox from what I hear.
There has been more noise about WebRTC making it possible to track users. We have covered some of the nefarious uses of WebRTC and look out for it before. After reading a blog post on this topic covering some allegedly new unaddressed issues a week ago I decided to ignore it after some discussion on the mozilla IRC channel. But this has some up on a the twitter-sphere again and Tsahi said ‘ouch’, here are my thoughts.
The blog post (available here) makes a number of claims about how certain Chrome behavior makes fingerprinting easier:
Chrome, Firefox, and Edge are all on the same party line. Image from Pillow Talk (1959)
For the first time, Chrome, Firefox and Edge can “talk” to each other via WebRTC and ORTC. Check the demo on Microsoft’s modern.ie testdrive.
tl;dr: don’t worry, audio works. codec interop issue…
| Feature | Interoperability | Notes |
| ICE | yes | Edge requires end-of-candidate signaling |
| DTLS | yes | |
| audio | yes | using G.722, Opus or G.711 codecs |
| video | no | standard H.264 is not supported in Edge yet |
| DataChannels | no | Edge does not support dataChannels |
As a reader of this blog, you probably know what WebRTC is but let me quote this:
WebRTC is a new set of technologies that brings clear crisp voice, sharp high-definition (HD) video and low-delay communication to the web browser.
In order to succeed, a web-based communications platform needs to work across browsers. Thanks to the work and participation of the W3C and IETF communities in developing the platform, Chrome and Firefox can now communicate by using standard technologies such as the Opus and VP8 codecs for audio and video, DTLS-SRTP for encryption, and ICE for networking.
We have been waiting a long time for Microsoft to add WebRTC to its browser portfolio. That day finally came last month when Microsoft announced its new Windows 10 Edge browser had ORTC. This certainly does not immediately address the Internet Explorer population and ORTC is still new to many (which is why we cover it often). On the positive side, interoperability between Edge, Chrome, and Firefox on the audio side was proven within days by multiple parties. Much of ORTC is finding its way into the WebRTC 1.0 specification and browser implementations.
So I talked about Skype and Viber at KrankyGeek two weeks ago. Watch the video on youtube or take a look at the slides. No “reports” or packet dumps to publish this time, mostly because it is very hard to draw conclusions from the results.
The VoIP services we have looked at so far which use the RTP protocol for transferring media. RTP uses a packet header which is not encrypted and contains a number of attributes such as the payload type (identifying the codec used), a synchronization source (which identifies the source of the stream), a sequence number and a timestamp. This allows routers to identify RTP packets and prioritize them. This also allows someone monitoring all network traffic (“Pervasive Monitoring“) to easily identify VoIP traffic. Or someone wiretapping your internet connection.
