Michael Welter wrote:
SER newbie here. Why do you need Asterisk for Sip->SIP setup? And if there is a reinvite, is that for the RTP stream only or for the SIP transactions as well? Will you lose the BYE transaction if there is a reinvite?Good questions.
Also, how many SIP registrations do you expect to maintain on each SER box?
I need asterisk for SIP->SIP setups because it can do "interesting" things with the call. Like capture digits, play prompts,listen for key codes during a live coversation to transfer and other neat things. As far as I know (and please correct me if I am wrong!) but SER can't do these kinds of things since it's totally disconnected from the RTP Stream.
As for the reinvite, I'm *hoping* that it's just the establishment of the RTP stream and not the entire signalling path. However I could be wrong. Looking for feedback from anyone who knows better.
I'm expecting SER to maintain "thousands" of registrations.
To quote the SER Admin manual:
"With a $3000 dual-CPU PC, the SIP Express Router is able to power IP telephony services in an area as large as the Bay Area during peak hours. Even on an IPAQ PDA, the server withstands 150 calls per second (CPS)!"
That is why you want SER.
The two together provide features + scalability.
I'm still not sure how to provide services that interact one phone with another phone's RTP stream. Like call pickup. How can I pickup a call on another asterisk server? Hmm Hmmmmmmmmm
-Brett
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