The above (SIP and H.323) was explaining the handshake between the ip routers over the internet. All internet connections are mostly not secure. Lost handshake or session packets must be retransmitted as long as the confirmation did arrive.

But the connection between (analogue or digital) voice switches inside a "Public Switched Telephone Network" (= PSTN) is based on much more reliable lines from copper or fiber optics.

However, there must be a standard communication, we say signaling, between voice switches amongst semselves and peripherals like gateways.

A sample:

The ISDN signaling protocol

If you run an ISDN dial in router" with one E1 WAN port (2 Mbit/s), you must configure channel 0 and channel 16 as signaling channels and channel 1 to 15 and 17 to 31 as data channels. (this is on a MAX 4000 or 6000) That means, from your 32 channels you are loosing 2 channels for ISDN signaling. If you run a T1 dialin line (1.5 Mbit/s US only), you have 26 data chanels from 28 channels at all. It does not matter the 64k and 56k difference in bandwidth.

So each E1 port must do its handshake for all 30 channels itself. Thats a lot of communication- and processor- load inside the PSTN voice switch.

The SS7 signaling protocol

is using a full E1 or T1 WAN port for signaling. It makes no difference, how many E1 ports are connected. Having an 8port E1 card in a TNT only, ist seems to be inefficient, but it isnt. SS7 must be faster. The more E1 ports you are using, the less ist the supposed loss of the one SS7 port.

The price makes the difference too:

Much more important ist the difference in price, when the telco offers much better rates per E1 port per month. They can use much cheaper equipment with standard telco protocol (SS7 ist the standard).

So you should request both offers for E1/T1 PRI (primary rate interface) ports conected to the telcos voice switches.

We hope, this explanation helps a lot.