Asked this on SIP forum also but no response so asking here. Please help
I've been reading about Voice Over IP and it's still not clear to me which protocols (i.e. RFCs) you have to implement to support audio calls between two IP-endpoints where the two endpoints might not reside on the same network.
So far I understand that:
- RTP (RFC 3550) is used for audio payload transport
- SIP (RFC 3261) is used for signaling
- SDP (RFC 4566) is used for media negotiation (i.e. which codec to use)
- RTCP (RFC 3550) is used for transmission stats
So if I'm not mistaken it should be possible to establish a SIP audio call between two devices with an implementation of the 3 above mentioned protocols (?)
But what I'm not totally sure about is :
- Which RFCs do you need to implement to ensure that a SIP audio call can be established between two devices which reside on different (firewalled) networks ?
- How do SIP phones typically get around network issues where two SIP clients are on two totally different networks and each client is behind a firewall? I assume that traffic to certain ports on a public IP address MUST be broadcasted to local listeners on those ports? But is that enough?
What if port forwarding isn't allowed on the network where a SIP client is residing?
