[Asterisk-Users] delay problem in h323

[andy]

yes, I agree with you.
I verify with a sniffer and asterisk manages RTP flows. The problem is asterisk 
decode and then code again RTP flows. This function requires 5-7% CPU On my 
test-box (Linux rh 7.3 on P3 600 GHz). This solution  don't scale without dedicated 
HW, I think!

Another problem is codec supported: ok for G.711, G.729. I don't know for GSM 
BUT: what about video codec? what about proprietary codec or ciphered codec?

Do you have any suggestion on how I can manage this with asterisk? I'm very 
interested into asterisk as sip-to-h323 translator.
Thanks

Andrea


Quoting Steven Thomas <vcsteven at au1.ibm.com>:

> 
> 
> 
> 
> 
> The only way I was able to solve my delay issue with Chan_oh323 was to
> switch to Chan_h323.
> 
> Chan_oh323 caused a similar 3 -4 sec delay on one way of the conversation.
> Checking the CPU stats on asterisk during the call - confirms that the RTP
> stream was somehow routing through asterisk - not sure why!
> 
> 
> 
> Regards,
> 
> Steven Thomas
> 
> 
> 
> 
>                                                                              
>                                                                
>                       andrea <andrea at csp.it>                                 
>                                                                
>                       Sent by:                          To:      
> asterisk-users at lists.digium.com                                            
>                       asterisk-users-admin at lists        cc:                  
>                                                                
>                       .digium.com                       Subject:  Re:
> [Asterisk-Users] delay problem in h323                                 
>                                                                              
>                                                                
>                                                                              
>                                                                
>                       10-09-03 12:45 AM                                      
>                                                                
>                       Please respond to                                      
>                                                                
>                       asterisk-users                                         
>                                                                
>                                                                              
>                                                                
> 
> 
> 
> Hi all,
> 
> is it possible to disable RTP routing through asterisk? RTP routing is a
> very nice feature but, I think it’s important also to disable it in some
> cases (e. g. in a LAN).
> Do you have any suggestion?
> 
> Andrea
> 
> Rattana BIV wrote:
> 
> > Hi,
> >
> > I have a delay between two H323.
> >
> > Netmeeting1 --------- |            |
> >                              | gnuGK | ----------- [asterisk-oh323]----
> > | Asterisk |
> > Netmeeting2 ----------|            |
> >
> > Netmmeting 1 call Netmeeting 2. When Netmmeting 1 speak Netmeeting 2
> > receive the voice without delay. But in the other way I have 3 secondes
> > delay.
> > In oh323.conf  I set jittermin and jittermax to 20, the ipTos=lowdelay.
> > I try to find where I can delete the delay.
> > Does anyone have a tip ?
> >
> >
> > Best Regards
> > Rattana
> >
> 
> 
> 
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