In Cisco UCM solution, there is a mechanism called Call Admission Control (CAC).
The CAC always be performed during call setup phase. If I was not wrong, there are two type CAC: 1. Topology unaware CAC 2. Topology aware CAC I don't think the second type CAC can be implemented with sipXecs, because the IP PBX (call manager) have to communicate with Cisco network devices to reserved the bandwidth (using RSVP protocol). But I think the first option can be implemented with sipXecs, cause it's only depend with the IP PBX and the configuration inside the IP PBX. I think it will be a very great features if sipXecs support this kind of mechanism. The information for CAC can be read/download here (http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/7x/uc7_0.html) From: Alberto [mailto:[email protected]] Sent: Thursday, June 25, 2009 4:14 PM To: Boy Aidil Sjam Cc: [email protected] Subject: Re: [sipx-users] Limiting voice call session Hi, you currently focused this issue better than I did. I thought about the bandwidth management as sipxbridge problem, but definitely I realize is more general than that. There is a JIRA issue to address call limit in sipxbridge: http://track.sipfoundry.org/browse/XX-5871 The interesting question you spotted is: how could we generally limit Internet calls and not just sipxbridge calls? One single limit for all Internet calls is enough, or is better to have a per gateway limit as well? I'm just thinking at a deployment were calls could be routed on different WANs based on the itsp / remote sipxecs used. Alberto Boy Aidil Sjam ha scritto: Hi All, Let say, I have two sipXecs in different domain or one sipXecs server connected to other SIP server (ITSP) or gateway. The connection between both location is separated with WAN connection with limited bandwidth to share between the voice call and others data application (256kbps). I had setup QoS between two network location. For VoIP I guarantee maximum 2 voice call at the same time with G.711 codec (about 160 kbps). But when a heavy call time, the call between 2 sites can rise up to 5 voice call at the same time. And when this happen the voice call quality will start to degrade. Is there any mechanism in sipXecs to limit voice call between two sipXecs or to other gateway, so when the 3rd call start to dial, it will reply with busy? Regards, B. Aidil DISCLAIMER: Information in this message is confidential and may be legally privileged. It is intended solely for the addressees. Access to this message by anyone else is unauthorised. If you are not the intended recipient, any disclosure, copying, or distribution of the message, or any action or omission taken by you in reliance on it, is prohibited and may be unlawful. Please immediately contact the sender if you have received this message in error. _____ _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users sipXecs IP PBX -- http://www.sipfoundry.org/ Checked by AVG - www.avg.com Version: 8.5.374 / Virus Database: 270.12.90/2200 - Release Date: 06/24/09 12:49:00 DISCLAIMER: Information in this message is confidential and may be legally privileged. It is intended solely for the addressees. Access to this message by anyone else is unauthorised. If you are not the intended recipient, any disclosure, copying, or distribution of the message, or any action or omission taken by you in reliance on it, is prohibited and may be unlawful. Please immediately contact the sender if you have received this message in error.
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