Video calls, from PortaSIP perspective, are very similar in flow to the conventional (voice only) calls described in the Call Process/Supported Services section of this guide). In video calls, however, there are multiple RTP streams: for audio and video.
SIP signaling flows between end-point and PortaSIP (and PortaSIP performs call validation using PortaBilling via RADIUS protocol) – in exactly the same manner as it does for voice calls. This allows to control the authorization, authentication, and call flow in accordance with the settings and balance of the account.
Just like a voice call, the RTP streams can go directly from one video end-point to another or be mediated by RTP proxy, if necessary (for instance both end-points are on separate private networks behind NAT). The main considerations for providing video call service are the following:
- End-points (IP video phones or communication clients) involved need to support video calls (some supported models are Hardware phones: Polycom VVX 1500, Grandstream GXV3140, Grandstream GXV3175; Softphones: eyeBeam, X-Lite, Ekiga).
- In case the call goes to or from PSTN, the gateway should be able to process video calls, too.
- Due to the much higher required bandwidth usually it is advisable to provide video calls only to clients on public IPs, so the RTP streams can be connected directly and no proxying on PortaSIP side is required.