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SIP Concepts – Record Routing

Explanation of Record-Routing, why it exists and how it’s used.

SIP was designed to be flexible in it’s operation, and for, where possible, messages to take the most direct path.

For example I can use a Registrar function of a proxy to find the IP of a registered endpoint, but once a dialog is setup, why should the proxy be involved? The endpoint & I can take it from here, and can talk directly to each other using the address in the Contact header.

This works really well in some scenarios, as I described above you can have the registrar proxy setup the introduction and then off you go.

Other scenarios this doesn’t work quite so well, for example if the call needs to be billed. To charge correctly, the proxy needs to know when the call ends to know when to stop charging.

If the endpoint we’re talking to is behind a NAT, the NAT might just be locked to the IP of the registrar proxy and drop your traffic.

The Record-Route header exists to address this.

If a proxy adds a Record-Route header, it means it’ll sit in the middle of any future requests in this dialog, and route them back through the proxy.

By adding a Record-Route header on the proxy for our billing example, our proxy will forward inline all the messages between the two end points for that dialog, including the BYE so the proxy knows when to stop charging.

For the NAT scenario we described the Proxy will add a Record-Route header and forward all the messages between the two endpoints, so NAT won’t be an issue as the source IP of the packets will be the same as the proxy.

There was a bit of confusion in regards to implementation so to address this IETF wrote RFC 5658 to address Record-Route Issues in SIP.

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