SARK UCS/MVP UDP Port Issues

Overview

IP Ports

You will notice that SIP can potentially use a large range of ports. In simple terms, how it works is this; the initial call is set up using the 5060 control channel. During set-up, each partner chooses a transmit port (usually at random) from the available RTP ports and these are then used to carry the actual data packets during the resulting conversation, one for each "direction". This has huge implications for VOIP servers running behind NAT'ed firewalls because we potentially have to open and forward all of these RTP ports to the server since we can't predict in advance what inbound SIP port(s) may be chosen. Asterisk has a mechanism to reduce the size of the RTP Port pool and, by default, it narrows it to ports 10000-20000. However that's still a lot of ports! As a minimum, you will need a router which can port-range forward, otherwise you are going to have to spend a small lifetime opening and forwarding the necessary ports. As if that wasn't bad enough, you may wish to support remote co-workers on your exchange. Their SIP telephones may well be behind another Nat'ed firewall at a remote site. In this scenario it is possible that the two peers are so well protected by their respective firewalls that they can't even find one another, let alone set up a conversation!

IAX2, on the other hand uses a single port (4569) for all operations. By simply opening and forwarding 4569 to the server you can commence VOIP operations with any carrier who supports IAX2 for outbound and inbound calls.

IP Ports and their effect on Telephony

Server-Gateway Mode

In Server-Gateway mode, SARK UCS/MVP sets up the SME server database to open UDP ports 4569 (IAX2), 5060 (SIP control channel) and the SIP RTP ports 10000:20000. For most modern carriers (who will almost certainly be running SessionBorderControl), it may not be necessary to open the RTP ports and just having 5060 open will be sufficient to support full inbound and outbound SIP communications. Where you will have difficulty is if you want to support a remote SIP device which is behind its own firewall.

Server-only Mode

Ensure that you put the external ip address of your site into the Globals panel in SARK UCS/MVP , otherwise you may get one-way sound on SIP operations.