[Asterisk-Users] delay problem in h323

Wed Sep 10 03:53:35 MST 2003

I assume it manages the signal part of the RTP stream but not the RTP voice
stream at the codec level?

Maybe someone else can comment on the translation methodologies within
Asterisk?

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 04:45 PM                                                                                                      
                      Please respond to                                                                                                      
                      asterisk-users                                                                                                         

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 itâ€™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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