[Spce-user] Packet Routing by SIP server

[Andreas Granig]

Hi,

On 03/28/2013 08:24 AM, lamarana jallow wrote:
> i am actually concerned about the individual packets that is sent from
> one UA to another.
> Lets say  after the connection has been established by the SIP provider,
> and the ACK has been
> sent to the callee and the media is now between the two UA's   what
> happens to the packets, do
> they go via the SIP server or not? So in this case when the call has
> been established  and the server is down
> does it mean that the call will drop?

Well, depends :)

If you mean the media streams, then on the SPCE there are user 
preferences "always_use_rtpproxy" and "never_use_rtpproxy" for 
subscriber/domain/peer to control whether the media relay on the SPCE 
will be engaged or not. If not, your media goes end-to-end (if you're 
not behind nat), so it won't matter if the SPCE goes down for a short 
time during a call.

However, UAs also (can) do session keep-alives using session timers on 
SIP level during a call, and if the SPCE is down, the UAs will end the 
call if they are not successful.

And finally, if you want to rely on correct billing, you want your SPCE 
to be always up to write an accounting record for the BYE, otherwise you 
won't get a CDR from these calls and can't bill your users.

Andreas