[Asterisk-Dev] How to make RTP Packets NOT passing thru Asterisk?

Partha Sarathi partha_sarathi_r at yahoo.co.in
Sat Aug 21 05:11:43 MST 2004


Hello All,

Thanks for your suggestion.I had changed my sip.conf 
as your advice and also check the link 
www.voip-info.org. My asterisk server configuration
follows all the rules correctly.sometimes xlite send
their RTP packets to destination without servers
interaction but sometimes it sill goes through the
Asterisk.In the case of sip-communicator(user agent) 
it transmits the both RTP and H.263 packets through
asterisk only. I have to change anything else? I am
new to the asterisk.Could you help me to solve the
problem.

please let me know you need any further information. 
Regards,
Partha
  
  


----- Original Message ----- 
From: "Simone Ricci" <simone.ricci at cwnet.it>
To: <asterisk-users at lists.digium.com>
Sent: Wednesday, August 18, 2004 8:07 PM
Subject: Re: [Asterisk-Users] How to make RTP Packets
NOT passing thru Asterisk?


> > Hello All,
> > 
> > Currently my setup uses Xlite and Asterisk and i
found that all the RTP
> > voice packets are transfered via the asterisk
server from one xlite to
> > another. Is there any possibility that we can make
all the RTP Packets to be
> > transfered directly between the two clients once
the connection is
> > established?.
> 
> 
> Simply use canreinvite=yes in peers definition. But
be sure to not use t 
> or T options in the dial command, because this
forces asterisk to care 
> about rtp streams (to manage transfers).
> BTW, you'll find this and others useful infos on
www.voip-info.org, 
> under asterisk section. Please read it carefully
before posting in 
> mailing list, thanks.
> 
> Cheers,
> Simone.
> _______________________________________________
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