WebRTC Testing Description and Call For Participation

IMTC SuperOp! 2015

Porto, Portugal

May 16 – 22, 2015

WebRTC Testing Description and Call For Participation

The IMTC WebRTC Activity Group (WebRTC AG) will be at the upcoming IMTC SuperOp to perform WebRTC interoperability testing. Due to the unspecified nature of WebRTC call control signalling, the IMTC SuperOp will focus testing on interoperability of WebRTC implementations to SIP with WebRTC media either converted to a more traditional SIP (SDES or unencrypted RTP) profile, or for additional scenarios a WebRTC based profile (DTLS-SRTP and ICE).  An example of such a scenario is the Sonus SuperOp environment shown below.

post1

In this environment Sonus provides the WebRTC GW and SBC. Any WebRTC compatible browser can initiate a session from a suitable webpage. The Call Control signalling connects to the Sonus WebRTC GW, where it converted into a vanilla SIP. This can then (via the SBC) be interworked to a range of SIP implementations, making it suitable for interoperability with enterprise or carrier focused SIP servers/solutions.

The media passes from the browser direct to the Sonus SBC. The SBC can then interwork that media to plain (unencrypted) RTP, a more traditional SDES based SRTP or “pass through” the DTLS-SRTP untouched. Audio and Video calls are supported, as well as range of transport and IP versions.  In other environments it may be possible to perform direct media interoperability between the WebRTC endpoint and the SIP endpoint if the SIP implementation supports WebRTC compatible media.

Another similar SuperOp test scenario is proposed by Unify in which SIP implementations with various levels of media compatibility with WebRTC are able to join collaboration sessions with Unify’s WebRTC based clients.  In fact in this scenario the Sonus WebRTC endpoints will be able to test interworking with Unify WebRTC endpoints using SIP for signalling and various combinations of media profiles (DTLS-SRTP, SDES, RTP).

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Given this brief description of the classes of interoperability supported by this configuration, the IMTC WebRTC Activity Group is interested in discussing specific scenarios other parties maybe interested in testing. This could be vendors with predominately SIP based applications looking to interop, or other WebRTC implementations that would be interested in using SIP as a common denominator to connect back-to-back.

Components of particular interest on the right hand side of the diagram would include MRF/video-transcoding entities, SIP endpoints, call servers, PBX’s etc. Additionally (not shown) we would also be interested in learning more about the behaviour of STUN/TURN and NAT/Firewall implementations that may operate in the user/browser space (in front of the Gateway & SBC).

 

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