Ronald Voermans wrote: > Hello, > > We are using OpenSer with RTPProxy for serveral years now. We recently > changed the hardware on which OpenSer/RTPProxy is running, and upgraded > all the software. > > I’m using OpenSer 1.3 with RTPProxy 1.1. > > We are forwarding all PSTN request to our ITSP. We are doing max. 40 > simultaneous calls. Since a couple of weeks our customers are > complaining that their calls are being dropped, and that there is a > significant amount of jitter. > > I have been looking for the cause of this for some time now, and have > come to the conclusion that it must be somewhere between the OpenSer and > our ITSP (have done serval monitoring on the customer site, and at our > site). Our ITSP says everything is running fine (as usual J), so I have > to come up with some evidence that the cause is indeed with the ITSP. > > First thing I would like to do is check if I’ve optimized everyting on > the (new) OpenSer server: > > The server is a HP Proliant DL140; a quadcore Intel processor (2.44 > GHz), with 4 GB of RAM. CPU is never above 0.1% and memory usage < 1 GB. > > I’m running only 1 instance of RTPProxy. In the FAQ’s I read that you > can better run 4 instances of RTPProxy. Is this also necessary with my > setup, and with the max. number of sim. calls I have? Will this have a > performance-gain if I run 4 instances? If so, could this be the cause of > the jitter/dropped calls?
Roland, I don't think there will be any improvements if you run several instances of the proxy in parallel with this amount of traffic. What could help a bit, though, is increasing size of kernel UDP buffers to avoid packet loss during traffic spikes, on Linux you can use the following commands: sysctl -w net.core.rmem_max=16777216 sysctl -w net.core.wmem_max=16777216 Commands above allocate 16MB to the send buffer and 16MB to the receive buffer. The default value is 128KB, which is too low for loss-sensitive VoIP. If it doesn't help, the way to investigate the issue is to capture network dumps either on the machine running proxy or even better on network's gateway. The tools like Wireshark (former Ethereal) could help you to analyze RTP streams given these captures. You can for example compare loss level and jitter for streams coming to the proxy and streams going out and show this data it to your ITSP. Just for your information we are currently working on feature that would gather real-time RTP loss/jitter statistics in rtpproxy. It will probably available in 2-3 weeks from now. Sincerely, -- Maksym Sobolyev Sippy Software, Inc. Internet Telephony (VoIP) Experts T/F: +1-646-651-1110 Web: http://www.sippysoft.com _______________________________________________ Users mailing list Users@rtpproxy.org http://lists.rtpproxy.org/mailman/listinfo/users
