Signaling Links: SIP vs. SS7

January 4th, 2007 by Emmanuel Buu

Traditional telephony links together SS7 switches using permanent, static connections. These are commonly referred as signaling links. The SS7 protocol (MTP2 and MTP3) has robust keep-alive and error correction mechanisms.

On the contrary, most of the SIP signaling links are done using UDP. So a lot of error correction and retransmission has to be done using the SIP protocol to account for the unreliability of the UDP protocol. SIP over TCP is possible but I would rather make it the standard. I would improve the RFC to impose permanent connection between user agents and registrar and whenever it is possible, make permanent TCP connection between SIP proxies. TCP keepalive should be mandatory for those connection. As a side effect, this would also help the NAT traversal for SIP.

For interdomain SIP communication, where the destination proxy is determined by a DNS SRV lookup, proxy to proxy connection needs to be made only when a SIP session between the two domains is initiated. I would make the connection linger and the reused in case of subsequent calls.

I dream of the day I can use SCTP for proxy to proxy connections (instead of TCP) but this is still a very “exotic” protocol.

One Response to “Signaling Links: SIP vs. SS7”

  1. Jonas B. Says:

    It’d be easier to take that suggestion seriously if you would counter the arguments that put SIP in UDP in the first place.

    One of the arguments is that answering a modern digital phone is very instant operation, from answer to first sound (”Hi..”) is a fraction of a second. Elimininating a whole round trip time makes the odds much better that the start of the sound actually gets through to the other end.

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