Sippy WebRTC Phone

What is it?

This is a technology demo integrating Sippy RTPProxy and Sippy B2BUA with WebRTC-compatible clients. It includes four main components:

1. Sippy B2BUA.
2. Sippy RTPProxy.
3. SIP.js demo application.
4. Web server.

How it works

The container starts RTPProxy and B2BUA listening on WSS port 9876/TCP, and a web server on HTTPS port 443/TCP. Both share the same self-signed TLS key generated during the container build process. This allows users to open the demo page and connect their browser to the B2BUA over WSS.

The only role of the HTTPS server is to allow user to download HTML/JS, it has no role in the actual real-time session so that particular component can be externalized.

Any registation attempts coming via the WSS socker are proxied to the external SIP registrar controlled by the OUTBOUND_PROXY environment variable via SIP/UDP.

When the user initiates a call, the B2BUA/RTPProxy sets up two RTP sessions (one encrypted and one plain) and initiates an outbound SIP call to the SIP destination controlled by the OUTBOUND_ROUTE environment variable. See Call Routing section of the Sippy B2BUA documentation for this parameter format.

When no OUTBOUND_ROUTE is provided, OUTBOUND_PROXY will be used instead as the only route to attempt.

Usage

docker pull sippylabs/webrtc_phone:latest
docker run -it --name webrtc_phone -P --network=host \
 -e OUTBOUND_PROXY="sip.mypbx.net" -d sippylabs/webrtc_phone:latest

Introspection

The container produces various SIP/RTP/WSS logs that can be inspected using the docker logs command. The amount of RTP logs can be controlled by the RTPP_LOG_LEVEL environment variable. Possible values are DBUG, INFO, WARN, ERR, and CRIT (in decreasing order of verbosity).

Performance

With the current configuration a single container should be able to support up to 500 concurrent users fully utilizing up to 5-6 cores. If you try to use it in a performance-critical scenario make sure to supply RTPP_NODEBUG=1 when running the container.

Specific range of UDP ports allocated by the RTPProxy can be controlled by the MIN_RTP_PORT and MAX_RTP_PORT parameters. At this moment each session allocates 4 ports, so that the range should be at least expected maximum number of simultaneous sessions times 4.

Caveats and Limitations

• Connection to the WSS server will fail with error 1015 in Firefox. It works in Chrome and Microsoft Edge as long as the user accepts the security warning when opening the demo page. This is caused by the usage of the self-signed certificate.
• Only Demo 1 and Demo 2 works.
• Due to the need for a range of UDP ports for RTP sessions (2,000 by default), the usage of the host network is recommended.

Links and References

• RTPProxy @ GitHub
• Sippy B2BUA @ GitHub
• SIP.js @ GitHub
• Sources for this container @ GitHub