Hi
To investigate the issue I'm having with uni-directional sound, I'd like to make sure I understood things correctly as to how SIP and RTP work together to achieve VoIP conversations ((A = caller, B = callee, B proxy = proxy server used by the callee to register):
- Both SIP devices register with a proxy to make themselves known and reachable by connecting to their respective proxies, each by connecting to UDP 5060 (or checking out SRV records in a dynamic DNS)
- SIP Caller connects to UDP 5060 on callee's proxy and dials the extension on B proxy (eg. sip:101@remote_proxy.com), opens a port locally to exchange RTP voice data, and sends this information to the B proxy
- B proxy rings the extension
- B negotiates with its proxy by receiving A's address and RTP port, opens its own port for RTP, and sends i
- Once A and B know each other's port available for RTP, the actual voice conversation can begin.
In other words, SIP + RTC require one port on a proxy to register (usually 5060), one port on SIP clients to receive INVITEs (usually
5070), and one RTP on SIP clients to exchange voice data (any port will do)?