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

[qcorba]

Please check RFC5853 regarding SBC.

Mauritz Jameson <mjames2393 at gmail.com> 於 2014年9月2日星期二 寫道：

> 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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