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Blog Central

SIP, RTP, NAT and your Zephyr XStream

Posted by Clark Novak on Jul 20, 2011 10:12:00 AM

Zephyr Xstream is the world standard for ISDN remote codecs. But it can also do IP remotes, and as ISDN circuits become harder to obtain, many stations are choosing to use their Zephyr Xstream's IP capabilities, which use industry-standard SIP and RTP to make connections over public IP networks.

SIP is a connection protocol (Session Initiation Protocol). RTP is a transport protocol (Real-time Transport Protocol). In the case of broadcast remotes via IP, the order of events is that SIP is used to initiate the transfer, and once the connection is made, RTP then carries the audio. A Telos Z/IP ONE or Zephyr Xstream initiates a SIP connection, and using that protocol, negotiates a connection over which each end can send an RTP stream to the other end. Then the SIP connection ends, leaving only the two RTP audio streams.

If you have a direct IP connection between the two ends of your remote, no problem - your XStream will connect via SIP and RTP easily. But many stations have a Network Address Translation layer, or NAT, which can be problematic. A NAT, as defined in the IETF's RFC 1631, is a way for local networks to allow multiple internal connections to and from the Web while consuming only one local IP address. Useful as this can be when managing IP address blocks, it can unfortunately interfere with the remote connection from your Zephyr.

Z/IP ONE codecs can handle NATs with no problem, using our complimentary Z/IP Server service to register and locate your remote connection behind the NAT. But Zephyr Xstream doesn't use Z/IP Servers - so how do you make the connection?

Betcha Didn't Know You Could do That with Your ZX

Here's the trick: on the last Codec menu screen of your Zephyr Xstream, set the WAN IP field to match your network's public IP address - the IP that is presented to the world outside the NAT. Why? Using this value, SIP will negotiate for each side to send an RTP stream, so it needs to know what IP address to tell the other side to stream to. By default, this is the Zephyr's IP address - but if it's behind a NAT, that IP address isn't reachable by the far side. Filling the WAN IP field tells SIP to use that public IP address instead.

Finally, make sure your NAT router is set to forward the proper streaming ports to the Zephyr. Zephyr uses TCP 5060 and UDP 9150 by default; you’ll also need to make available TCP ports 24 and 308 available (used for session updating), port 11926 (used for listening) and ports 5060 and 5061 (used for SIP negotiation).

With just a few minutes of easy configuration,, you'll be doing IP remotes with your Zephyr Xstream. Remember, however, that a Zephyr XStream was designed to be an ISDN codec that does a bit of IP -- in this role, it's no substitute for a purpose-built IP codec like our Z/IP ONE, which has technology like ACT built in to maximize IP connection reliability and audio quality over the public Internet. But, if you're in a pinch and the purchase of a Z/IP ONE is down the road a bit, your Zephyr XStream can fill in nicely.

Topics: Telos Systems, IP Telephony, broadcast codec, sip