Configuring a Rtcomm Gateway
The Rtcomm Gateway adds the capability for connecting Session Initiation Protocol (SIP) with Rtcomm WebRTC endpoints for the exchange of audio and video streams.
About this task
The Rtcomm Gateway (GW) is useful when you require federating between the Rtcomm network and different vendors' networks. The other network can be a network of WebRTC endpoints, which is using a different method for signaling, or it can also be a different network of Voice over IP (VOIP) devices or even the Public switched telephone network (PSTN). Such federation is possible as long as the other network provides an edge GW element that supports the widely adopted SIP protocol.
The Rtcomm Gateway supports both Interactive Connectivity Establishment (ICE) for SIP (based on RFC 5245) and trickle ICE for SIP (based on the IETF draft). This draft is still marked as a "work in progress" so this implementation might change in the future in accordance to this draft progress.
Procedure
Example
<rtcomm messageServerHost="<brokerhostname>" messageServerPort="<brokerhostport>"
<gateway sipContainer="false" externalPR="1.2.23.2:5063" allowFromSipEndpointRef="webrtc2, webrtc"></gateway>
</rtcomm>
<sipEndpoint id="webrtc"></sipEndpoint>
<sipEndpoint id="webrtc2" sipTCPPort="5067" sipUDPPort="5067" sipTLSPort="5068" host="*"></sipEndpoint>