[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?
_______________________________________________
This is the SIP Forum discussion mailing list
TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion
Post to the list at discussion at sipforum.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://sipforum.org/pipermail/discussion/attachments/20140906/12a9b539/attachment-0002.html>
More information about the discussion
mailing list