Everything worked as expected with very small effort. For example:Ī = x - signaling - fb : * x - message app send : src, csrc, vc recv : srcĪzure Communication Services has a simple API. I doubt these have any impact and are probably inherited from other applications. There are also non-standard extensions present in the SDP. Bandwidth Estimation (BWE)įor bandwidth estimation it uses receiver side support (based on REMB) instead of the more modern and optimized sender side bandwidth estimation (based on Transport Feedback). It also reserves 50 ssrcs for each stream (1501, 1551…) and during the initial establishment of the call 8 remote streams are pre-allocated in the remote SDP for future participants. Some other details at RTP/RTCP level is the usage of bundle, rtcp-mux and rtcp-rsize that are used in most of the platforms too. You can see in the next capture from the sender parameters how it is configured to use H264 at 200kbps. Also at least in the examples I tested the bitrate was pretty low. ACS doesn’t include simulcast support to adapt the video quality to the needs of different participants in the room. It uses RTX retransmissions for reliability. This is a fragment of the SDP answer with the audio channel information:Ī = rtpmap : 101 telephone - event / 8000 That is really uncommon for WebRTC platforms but not as surprising given the need of PSTN interoperability and the reuse of existing Microsoft infrastructure. It will likely be removed at some point since the standard explicitly forbids SDES since it is less secure than the standard DTLS requirement. That is simpler and provides faster establishment but is only supported in Chrome. However, the SFU/Rooms the keys exchange uses SDES and not the standard DTLS protocol. The encryption is based on SRTP as required by WebRTC. Full ICE doesn’t provide many advantages but doesn’t have any negative impact either. That is not very common in SFUs with public IPs because it is harder to implement. Thanks also to Fippo for his help testing.įor the direct connection to the SFU, it uses typical ICE UDP candidates but also ICE TCP candidates in port 3478. Note I did reach out to the Azure Communications Team at Microsoft to give this a brief review for technical accuracy where they could comment. As you’ll see, this one has some interesting peculiarities too. Microsoft has had a long and very unique history with WebRTC, so we were extra curious to see how WebRTC was used as part of this new offer. Whenever a 1.6 trillion dollar company does a product launch it is generally a big deal, and particularly so to those of us who deal with communications API’s on a regular basis. If you go this far I assume you know what Microsoft Azure is. Make sure to check out some of his past webrtcHacks posts or check out his blog over at where he provides many unique insights in shorter form. Gustovo has a deep career in real time communications and has been intimately involved with WebRTC since its very early days. Fortunately one of our favorite authors, Gustavo Garcia Bernardo recently found the time to review the new Microsoft Azure Communications Service, He found some interesting results that we are happy to present here. In a sign of WebRTC’s success, that list has been getting much longer and we’re not keeping up. We have a long tradition here of analyzing major services that use WebRTC.
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