[Asterisk-Users] SIP QUESTION

[Siggi Langauf]

On Tue, 19 Aug 2003, Jorge Cisneros Flores wrote:

>   Is posible to make a call from site A to Site C, and my question is, the rtp data is from A to C or is from A to B to C
>
>
>    Site A                             Site B                                      Site C
>    ata186<-------->FW<--------->Asterisk<--------->FW<----------->ata186

That depends.
Initially, RTP data is sent via Asterisk (B). Asterisk will then try to
reinvite A and C so RTP is sent directly between them. If it succeeds, RTP
is stent only between A and C, if it doesn't, asterisk will act as a
proxy. If the reinvite succeeds but a firewall is blocking the direct RTP
stream, the call will get silent. You'll have to put "canreinvite=no" in
the sip.conf for at least one of the ata186 devices in that case, so it
doesn't even try...

Cheers,
	Siggi