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

Saurabh Shah saurabh.shah at matrixcomsec.com
Sat Sep 6 04:01:16 UTC 2014


Please find below RFCs designed for NAT. 

RFC 3581 - Symmetric Response Routing (rport) 
RFC 5389 – STUN & RFC 5766 – TURN OR 
RFC 5768 - Indicating Support for Interactive Connectivity Establishment (ICE) in the SIP & RFC 5245 - Interactive Connectivity Establishment (ICE): A Protocol for Network Address Translator (NAT) Traversal for Offer/Answer Protocols 


Port forwarding is not required in clients which are behind NAT if registration is enabled. 


Regards, 
Saurabh Shah 

----- Original Message -----

From: "Mauritz Jameson" <mjames2393 at gmail.com> 
To: discussion at sipforum.org 
Sent: Tuesday, September 2, 2014 1:00:48 AM 
Subject: [SIPForum-discussion] Minimum set of required protocols for SIP calls ? 




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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