[Asterisk-Users] RTP session traversing Asterisk server ...

[Juan Heriberto Brito Jiménez]

Yes, i've observed the same operation :|, Adam.
I've the last CVS Asterisk, and two softphones (Linphone 1.12 and X-lite
v2 last version), both with speex code active.
When i call from one to another ... ringing ok but ... when try to talk
... the Asterisk go crazy warming "out of memory"  (i installed the
speex-dev in the server)

Are there anybody who know what's happen?

Thanxs, Heri.

PD.: Sorry my bad english :)


El lun, 28-07-2003 a las 15:29, Low, Adam escribió:
> I've been reading up on the SIP and related (SDP/RTP) RFC's and as I would expect the RTP session should ideally be between the two end points of the call, in my case the AS5300 and the 7940 which are connected on the same VLAN as the Asterisk server.
> 
> When I sniff the packets on the VLAN I find that all RTP packets are being relayed by the Asterisk server causing increased load on the server and ultimately a higher latency between the two end points.
> 
> Is this a typical operation of Asterisk or is this possibly due to the fact that some of the phones (not those used in the tests) are running NAT and Asterisk relays all RTP packets ?
> 
> Adam
> 
> 
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