Robert,

Yes, this is normal.  To interwork between a WebRTC client and a standard SIP 
client you need a gateway.

This is for two reasons.

1. WebRTC uses the "SAVPF" RTP profile, while most SIP clients use the "AVP" 
RTP profile.  The "SAVPF" profile includes two additional features:
  - security (the "S" in "SAVPF") - the WebRTC RTP packets are encrypted using 
DTLS-SRTP
  - early feedback (the "F" in "SAVPF") - the clients send each other more 
information about RTP packet arrival, meaning that they can better adapt their 
transmission.
Unfortunately, SAVPF is not backwards-compatible with AVP - both clients must 
support the same profile.

2. Sometimes, your WebRTC client will support different audio codecs from your 
SIP client.

We don't have a gateway as part of Project Clearwater.

I hope that helps. 

Thanks,

Matt


From: Robert Chen [mailto:[email protected]] 
Sent: 01 April 2015 07:59
To: [email protected]; Matt Williams
Subject: WebRTC client working with SIP client

Matt,

I am running Clearwater for a while. I noticed that, it is OK for WebRTC client 
(Sipml5) to call WebRTC client, and SIP client to call SIP client. But when 
WebRTC client calls the SIP client, and vice versa, the call is rejected with 
un-support media type error. Is this normal, or did I configured something 
wrong?

Thanks.

Robert 
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