[Asterisk-Users] RTP session between two end users

[Erdem HAKİ]

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Eric Wieling
aka ManxPower
Sent: Tuesday, June 28, 2005 6:19 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] RTP session between two end users

Erdem HAKİ wrote:

> 
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Eric Wieling
> aka ManxPower
> Sent: Monday, June 27, 2005 8:32 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] RTP session between two end users
> 
> Erdem HAKİ wrote:
> 
> 
>>Is it possible that a RTP session between two end users  (so i want to use
>>asterisk as a signaling proxy and bypass RTP sessions)?
>>
>> 
>>
>>I used "canreinvite=yes" but it didn't work. 
>>
>>
>>Description from asterisk conf. File;
>>
>>(canreinvite=yes                ; allow RTP voice traffic to bypass
>>Asterisk)
> 
> 
> 
> It's sip.conf.  reinvites only work if the codec is the same for the 
> two endpoints and Asterisk does NOT have to listen for DTMF (no t or T 
> on the dial line, no meetme, etc.)
> 
> ***************
> We use same codec and don't use meetme etc...  So what else should i do?
>
>How are you determining if RTP audio is going thru Asterisk? 
>Remember, SIP signaling will always go thru Asterisk.

>Also do a "sip show channels" during a call to confirm that the codecs 
>are the same.

-- 
>Always do right. This will gratify some people and astonish the rest.
Mark Twain
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************************

Hi,

I determine signaling with ethereal and i am sure that both sides use the
same codec.

By the way, i searched forum again and i read something below;

> In wiki pages it is stated that The audio channels (RTP) may go directly 
> from phone to phone or may go through Asterisk's media bridge.
> 
> Currently with my settings, I notice that all rtps are passing through
>  my asterisk. How could I achieve that they go directly from phone to
> phone?  I assume this way, my machine will have less load and therefore 
> could handle more calls.

As bkw pointed out, use canreinvite=yes for each sip phone definition.
But, that will only work if the phones can reach each other directly
(the phones and/or asterisk can't be behind a nat/firewall box).



Thanks 

Erdem HAKI -erdemh at tesas.com