asterisk users - Apr 2006 - SIP conections, with RTP not going trough Asterisk

Hi,

   Ie been looking for some time how to use asterisk  to initiate SIP 
connections between 2 IP phones,  but afetr initiated the communication 
making the RTP go directly from one telephone to the other, without 
passing by asterisk. Unfortunately I found no explanations of how to do it.

Does anyone care to give a pointer to any explanation about how to do it?


Thanks in advance

Tiago Stein D`Agostini wrote:
> Hi,
>
>   Ie been looking for some time how to use asterisk  to initiate SIP 
> connections between 2 IP phones,  but afetr initiated the 
> communication making the RTP go directly from one telephone to the 
> other, without passing by asterisk. Unfortunately I found no 
> explanations of how to do it.
>
> Does anyone care to give a pointer to any explanation about how to do it?
>
canreinvite=yes
and look at the options for dial()
>
> Thanks in advance
>

bye

Ronald Wiplinger

Hi, sorry to bother again. But I still cannot make it work. I made all 
acounts have canreinvite=yes, but found no option in Dial aplication to 
make the phones exchange RTP directly between them.  Can anyone tell me 
wich option should I look at? I am stuck with this (probably simple) 
problem for almost a whole week.

Thanks for any help.

Ronald Wiplinger wrote:
> Tiago Stein D`Agostini wrote:
>
>> Hi,
>>
>>   Ie been looking for some time how to use asterisk  to initiate SIP 
>> connections between 2 IP phones,  but afetr initiated the 
>> communication making the RTP go directly from one telephone to the 
>> other, without passing by asterisk. Unfortunately I found no 
>> explanations of how to do it.
>>
>> Does anyone care to give a pointer to any explanation about how to do 
>> it?
>>
> canreinvite=yes
> and look at the options for dial()
>
>>
>> Thanks in advance
>>
>
>
> bye
>
> Ronald Wiplinger
> _______________________________________________
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>

Tiago Stein D`Agostini wrote:
> Hi, sorry to bother again. But I still cannot make it work. I made all 
> acounts have canreinvite=yes, but found no option in Dial aplication to 
> make the phones exchange RTP directly between them.  Can anyone tell me 
> wich option should I look at? I am stuck with this (probably simple) 
> problem for almost a whole week.
The canreinvite=yes is required, however your Dial statements used to 
complete calls between the sip devices cannot use several of the options 
including t, T, etc.

If you remove all options from the Dial statement, restart asterisk, and 
place a test call, those sip phones that can "see" each other will 
auto-negotiate rtp directly between them.

If they cannot see each other (eg, nat or firewalls involved), they will 
not auto-negotiate direct rtp.

There is no option for you to specify to "forced" direct rtp.

Hi Ronald!

Please check if the following points are NOT activated. 
* is not using direct phone to phone RTP streams if:

-) either of the clients is configured with canreinvite=no
-) the clients cannot agree on a common set of codecs and * needs to
perform codec conversion
-) either of the clients is configured with nat=yes
-) * needs to listen to DTMF tones during the call (for transfers or any
other features)

Hope this helps,
Alex

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tiago
Stein D`Agostini
Sent: Montag, 17. April 2006 13:13
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SIP conections,with RTP not going trough
Asterisk

Hi, sorry to bother again. But I still cannot make it work. I made all 
acounts have canreinvite=yes, but found no option in Dial aplication to 
make the phones exchange RTP directly between them.  Can anyone tell me 
wich option should I look at? I am stuck with this (probably simple) 
problem for almost a whole week.

Thanks for any help.

Ronald Wiplinger wrote:
> Tiago Stein D`Agostini wrote:
>
>> Hi,
>>
>>   Ie been looking for some time how to use asterisk  to initiate SIP 
>> connections between 2 IP phones,  but afetr initiated the 
>> communication making the RTP go directly from one telephone to the 
>> other, without passing by asterisk. Unfortunately I found no 
>> explanations of how to do it.
>>
>> Does anyone care to give a pointer to any explanation about how to do
>> it?
>>
> canreinvite=yes
> and look at the options for dial()
>
>>
>> Thanks in advance
>>
>
>
> bye
>
> Ronald Wiplinger
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
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Thanks, that was the problem,  I had the t option on the Dial 
application. Nor that I removed them  it works.

Thank you.

Rich Adamson wrote:
> Tiago Stein D`Agostini wrote:
>
>> Hi, sorry to bother again. But I still cannot make it work. I made 
>> all acounts have canreinvite=yes, but found no option in Dial 
>> aplication to make the phones exchange RTP directly between them.  
>> Can anyone tell me wich option should I look at? I am stuck with this 
>> (probably simple) problem for almost a whole week.
>
>
> The canreinvite=yes is required, however your Dial statements used to 
> complete calls between the sip devices cannot use several of the 
> options including t, T, etc.
>
> If you remove all options from the Dial statement, restart asterisk, 
> and place a test call, those sip phones that can "see" each other
will
> auto-negotiate rtp directly between them.
>
> If they cannot see each other (eg, nat or firewalls involved), they 
> will not auto-negotiate direct rtp.
>
> There is no option for you to specify to "forced" direct rtp.
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>