[SIPForum-discussion] Minimum set of required protocols for SIP calls ?

[Mauritz Jameson]

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?
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