As an example of the new capability, application can instantiate both the sound device and the bridge in stereo. If the call is established with a stereo codec (for stereo we only have L16 codec for now), then transmission will be stereo, and if the call is established with the usual mono codec (G.711, G.722, Speex, etc. And oh yes, we do have G.722 now!), then the bridge will correctly perform the stereo - mono conversion, mixing the audio channels as necessary.

Having said all that, you may ask, why? Why bother with stereo at all?

You're right, stereo is uncommon in VoIP. But nowadays SIP is used in broader industries than just VoIP. For example, the European Broadcasting Union (EBU) is currently looking to standardize Audio Contribution over IP (N/ACIP) which is based on standard based protocols such as SIP and RTP, and stereo encoding comes as one of the mandatory requirements. Also on different industry, we're currently looking to integrate PJSIP as a VoIP platform in Virtual World games, in the open source VoIP for Virtual World project. This project is still in its very early stage, as it's been published literally just couple of weeks ago. Here I guess stereo will play quite an important role, because of the positional audio requirement. And still there are more use cases for stereo, for example in SIP to radio gateways where people do want to transmit different audio in the left and right channel. The last use case probably is niche, but people do ask for that.

So in conclusion, stereo does have a use case. And I'm glad that we have a pretty smooth support for that.

And how can we enable stereo in the application? Well just set pjsua_media_config.channel_count to 2, and that's just about it! More advanced scenarios are possible of course, but at least it's very easy to get started with it.