[Asterisk-Users] delay problem in h323

[andrea]

thanks, I'll try. Question: asterisk always manages RTP flow also with 
chan_h323?

Andrea

Steven Thomas wrote:

> 
> 
> 
> 
> Hi,
> 
> I use Asterisk as a SIP <-> H323 translator without any issues after
> switching to chan_h323.
> 
> My environment is:
> 
> SIP (7960) -> Asterisk -> GnuGK (h323) -> Cisco 2600 H323 Gateway to PSTN.
> 
> This works well without the CPU load seen with oh323.  The call control
> also seems far better using chan_h323.  I have no delay either.
> 
> I use a smaller box: PII 200, 64Mb RAM.  RedHat 9.
> 
> Only 4 handsets - 2 SIP IP 7960's, 2 Analog H323 via the Cisco router FXS
> ports.
> 
> I also have configured Asterisk on another site to act as a H323 gateway
> for PSTN calls into a Cisco Call Manager via gnuGK - H323 also.
> 
> I would suggest trying chan_h323 as an alternative.....
> 
> 
> 
> Regards,
> 
> Steven Thomas
> 
> 
> Technical Project Manager
> Network & Connectivity  Services, IBM Australia
> 
> Ph: 0404 099 262
> NH011, IBM Centre, St Leonards, 2065
> Internet:  vcsteven at au1.ibm.com
> 
> Visit us at http://www.ibm.com/services/au/its
> 
> 
> 
>                                                                                                                                              
>                       andy <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 08:24 AM                                                                                                      
>                       Please respond to                                                                                                      
>                       asterisk-users                                                                                                         
>                                                                                                                                              
> 
> 
> 
> 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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