[Asterisk-Users] delay problem in h323

[Steven Thomas]

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                                                                                                      
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                      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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