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: [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
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