[asterisk-biz] RTP redirect

[Ibs Bis]

check in : http://www.voip-info.org/wiki-Asterisk+SIP+canreinvite
  Asterisk sip.conf, peer definition: canreinvite option      
  This peer option in sip.conf is used to tell the Asterisk  server to never issue a reinvite to the client. This is used to  interoperate with some (buggy) hardware that crashes if we reinvite,  such as the common Cisco ATA 186.  
    
  When SIP initiates the call, the INVITE message contains the  information on where to send the media streams. Asterisk uses itself as  the end-points of media streams when setting up the call. Once the call  has been accepted, Asterisk sends another (re)INVITE message to the  clients with the information necessary to have the two clients send the  media streams directly to each other.  
    
    
    If one of the clients is configured with canreinvite=NO, Asterisk will not issue a re-invite at all.  
    If the clients use different codecs, Asterisk will not issue a re-invite.  
    If the Dial() command contains t, ''T", "h", "H", "w", "W" or "L" (with multiple arguments) Asterisk will not issue a re-invite.    
    
  'canreinvite=no' stops the sending of the (re)INVITEs once  the call is established. From messages in the archives and the Asterisk  handbook one finds out that the Cisco ATA-186 does not handle the  (re)INVITE well. This is necessary if the client and the Asterisk  server is on opposite sides of a NAT gateway or firewall.  
    
  Notes      
    reinvite=yes/no is plain wrong, even if you see it mentioned in example .conf files. The correct syntax is canreinvite=yes/no  
    Connecting media paths direct to an endpoint behind NAT won't  be pretty. Especially if both devices are behind NAT. You might want to  try using SER's nathelper in conjunction since nathelper.so can rewrite  the SDP so that the private IP addresses are not included in the  re-invite. ....
    
  

support <brian at ezzitel.com> a écrit :  So you want asterisk not to proxy the media..  Simple.  Put
canreinvite=yes in your peer in sip.conf.  It will do it automatically.



On Fri, 2006-02-17 at 01:19 +1100, Zafer Khodr wrote:
> I am trying to setup asterisk as shown below.
> 
> 
>      
>                         xxx.xxx.xxx.xxx
>        /=====(Call Origination)====\
>       /        |               \
>         R /        |S               \
>        T /   |I                \ R
>       P /   |P                 \ T
>         /   |                   \ P
>     S /     Asterisk                \ 
>    T /              /\                    \ S
>   R /        /  \                    \ T
>  E /            /    \                    \ R
>      A /                   /      \                    \ E
>     M /                   /        \                    \ A
>      /                   /          \                    \ M
>     /                   /            \                    \
>    /                   /              \                    \
>   /         __________/                \__________          \
>  /         |             |          \
> /      |             |           \
> yyy.yyy.yyy.yyy         zzz.zzz.zzz.zzz
> Terminating Gateway 1     Terminating Gateway 2
> 
> 
> 
> The main objective to to have asterisk in the path of the call but the RTP
> to go directly between the originating and terminating IP's.
> I have had a play around with canreinvite but doesn't seem to make a
> difference.
> 
> If someone could please help me out by pointing me in the right direction,
> that would be great
> 
> Regards,
> Zafer
> 
> 
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> 
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