[Asterisk-Users] How does call routing actually work with SIP?

[Kevin Walsh]

Daryll Strauss [daryll at daryll.net] wrote:
> As I said, I'm testing on a server I do use for other things. I plan to
> dedicate one before this goes live, but it lets me experiment. Yesterday
> I ran a big file transfer on my server and had a friend call me at the
> same time. Although his voice sounded fine to me, he heard a "worble" in
> my voice. I'm sure that's caused by traffic load I had on my server.
>
That can be caused by the lack of QoS on your network.  You need to
give priority to VoIP traffic - especially if you are saturating your
network connectivity.

> 
> That would seem to imply that the data packets continue to go through my
> Asterisk server after the call terminates on my phone. Is that true?
> Since this call came in on the Sipura PSTN and ended on the phone
> connected to the Sipura, it could have stayed within the Sipura box all
> together. 
> 
> Does SIP have a way to tell the originator to transfer the call to
> another SIP device? Is that something I can tell Asterisk to do in this
> case? It would be nice if two internal callers could talk without having
> the data go through the Asterisk. Then you'd have a nice switched
> network instead of a store and forward through a single node.
> 
You are looking for "canreinvite = no", which is a sip.conf setting.
This will allow the endpoints to establish a direct link to one another,
and remove Asterisk from the loop.  Of course, you can only remove
Asterisk from the loop if Asterisk is no longer required.

Asterisk will remain in the loop if you have specified "t" or "T" in
your Dial() command, as it will need to listen for the hash key.  It
will also remain in the loop if you're recording the audio stream
using Monitor(), or whatever.

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