asterisk users - Dec 2003 - caninvite...

hi guys

just got a question, im using grandstream phones with canreinvite=no or woteva,
all nat etc is working perfectly. but i believe because of the canreinvite, when
a call has taken place the voice will be proxied via the sip server to the 2
parties involved. ( which means the sip server is downloading/uploading to each
party constantly). Im just curious though with this setup for all clients.. so
everything goes through the sip server, how many phone calls do you rekon
asterisk could handle if it was say dual 2g or something like that ? I think i
read somewere else it was like 60-90 i forget.. but i think that was if rtp was
being handled properly.


Thanks heaps guys

Justin

vocalvoip wrote:
>hi guys
>
>just got a question, im using grandstream phones with canreinvite=no or
woteva, all nat etc is working perfectly. but i believe because of the
canreinvite, when a call has taken place the voice will be proxied via the sip
server to the 2 parties involved. ( which means the sip server is
downloading/uploading to each party constantly). Im just curious though with
this setup for all clients.. so everything goes through the sip server, how many
phone calls do you rekon asterisk could handle if it was say dual 2g or
something like that ? I think i read somewere else it was like 60-90 i forget..
but i think that was if rtp was being handled properly.
>
>
>Thanks heaps guys
>
>Justin
>  
>
Yes that is true, All traffic will go via the Asterisk server..

As for how many channels, this would depend on your codec.. you would be 
able to handle far more channels using G.711 than if you were using GSM 
or G.729..

Later..