[SIPForum-discussion] IMTC SuperOp! 2015 - WebRTC Testing Description and Call For Participation

Marc Robins marc.robins at sipforum.org
Tue Apr 14 19:17:31 UTC 2015

IMTC SuperOp! 2015

Porto, Portugal

May 16 - 22, 2015

WebRTC Testing Description and Call For Participation



The IMTC WebRTC Activity Group will at the upcoming IMTC SuperOp performing
WebRTC interoperability testing.  Due to the unspecified nature of WebRTC
call control signaling, 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.


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 signaling connects to the Sonus WebRTC GW, where it is 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 directly 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 a 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 signaling and various
combinations of media profiles (DTLS-SRTP, SDES, RTP).


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 behavior of STUN/TURN and NAT/Firewall implementations that
may operate in the user/browser space (in front of the Gateway & SBC).


Call for Action:


.       IMTC members interested in participation in WebRTC testing should
register directly at http://www.imtc.org/event/superop_2015/

.       Non-members, please contact Nathalie Mariano at
nmariano at inventures.com to inquire about participation




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