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: [EMAIL PROTECTED]
Visit us at http://www.ibm.com/services/au/its
andy <[EMAIL PROTECTED]>
Sent by: To: "" <[EMAIL
PROTECTED]>
[EMAIL PROTECTED] 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 <[EMAIL PROTECTED]>:
>
>
>
>
>
> 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 <[EMAIL PROTECTED]>
>
> Sent by: To:
> [EMAIL PROTECTED]
> [EMAIL PROTECTED] 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
> >
>
>
>
> _______________________________________________
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
> Ë^®+$RÇ«²f¢–)à–+- Ë^®+$RÇ«²X¬¶Çb‚+Šr‰¡¶ÚþX¬¶Çb‚+Šr‰¿™š¥™©ÿ–+-Šwèý«-
zž¬’ë ®
---------------------------------------------------
This mail sent through CSP Webmail System
_______________________________________________
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
,���E,z�&j)b� b��,���E,z�%��lv("�g(�m����lv("�g(����Y���b�ا~�ڲ��.���
