Re: [asterisk-users] Diagnosing poor call quality
Check from the sites in question using testmyvoip.com or whatever the site is called. In the UK I found that some strange things sometimes happen. At one point I was sure that BT were perhaps misclassifying IAX packets as P2P... However, not had a problem with SIP. Beware that ADSL uses vastly more bandwidth than you expect on small packets, eg if you are classifying using a cheap router then you probably need to at least half your claimed bandwidth in order to make the prioritisation work correctly. I added some (hack) patches to fix the linux calculation for HTB on the linux QOS list a year or two back. If you have a linux router you could use those to improve the calculation quality for QOS - or else I found a Draytek router does impressively well at getting it right for small sites... Very likely you will find that the issue is variable jitter on the line. The link above should help you figure this out Good luck Ed W ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Diagnosing poor call quality
Beware that ADSL uses vastly more bandwidth than you expect on small packets, eg if you are classifying using a cheap router then you probably need to at least half your claimed bandwidth in order to make the prioritisation work correctly. I added some (hack) patches to fix the linux calculation for HTB on the linux QOS list a year or two back. If you have a linux router you could use those to improve the calculation quality for QOS - or else I found a Draytek router does impressively well at getting it right for small sites... Each site is using an old PIII-era PC running m0n0wall (www.m0n0.ch/wall) as a router. Network cards are all decent-quality Intel Pro/100+ cards. I've not had any complaints from either site today - I made the RTP change on the SPA942s yesterday, and last night changed some of the QoS settings on m0n0wall at the ip290 site. Very likely you will find that the issue is variable jitter on the line. The link above should help you figure this out If this is the case, would upgrading to 1.4 with the new SIP jitter buffer help at all? Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited This email is made from 100% recycled electrons ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Diagnosing poor call quality
Chris Bagnall wrote: Greetings list, We have an issue with call quality at 2 sites where the users (4 Elmeg IP290s at one site, 2 SPA942s at the other) do not have an asterisk box on-site. Each site has an 8mb down/448k up ADSL connection and the phones connect via SIP to an asterisk box in a datacentre using g729. The asterisk box in the datacentre connects to our other asterisk boxes providing pstn connectivity via IAX2. Latency between these boxes is between 1 and 2ms. The ADSL connections to the client sites are all consistently delivering latencies of sub-25ms to the datacentre and there is traffic shaping on that connection to give priority to any traffic from the phones' IPs. Comments from the users at these sites are as follows: call sounded like a dalek and I couldn't make out anything the caller was saying the phone on my desk is breaking up so badly it's virtually unusable calls sound like they're breaking up with metallic background noises We have quite a few customers with asterisk boxes on-site (with phones connected to them via the LAN) using ADSL connections from the same supplier, and are not having these issues with them. canreinvite=no and nat=yes are set on all these devices, since they are behind NAT. Each device re-registers with asterisk every 5 minutes to prevent any possible NAT state timeouts. Any pointers/places to look for potential problems would be much appreciated. This should be a FAQ. Set the RTP packet size on the SPAs to .2 instead of .3 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Diagnosing poor call quality
The advertised datarate (8mb/448k) are the speeds at which the circuit between the customer and the central office is clocked and has no relationship with *effective* throughput. At the central office are *shared* facilities than connects each DSL connection with the network, and over subscription to these shared facilities cause congestion. Also, there is no QoS on the Internet, and congestion anywhere between the end points will cause poor call quality. Disclaimer: The following information is several months old--I've since moved my customers away from Qwest DSL. Here in Denver we have Qwest DSL service from a central office where the effective throughput drops to dialup speeds during the day. Regular web/email users don't usually notice packet loss because dropped packets are recovered by the TCP protocol. For VoIP on UDP, however, the call quality suffers to the point of being unusable (clicking, popping, and dropouts). Furthermore, Qwest doesn't have Denver peering with the rest of the Internet. To leave the Qwest network, connections typically go to DAL, LAX, or SFO on congested circuits. So beware of VoIP over DSL. Your users need to be aware of the tradeoffs between the cost of DSL vs. T1 and the effect on call quality. Chris, if your customers are in the western US then please contact me about dedicated circuits. Chris Bagnall wrote: Greetings list, We have an issue with call quality at 2 sites where the users (4 Elmeg IP290s at one site, 2 SPA942s at the other) do not have an asterisk box on-site. Each site has an 8mb down/448k up ADSL connection and the phones connect via SIP to an asterisk box in a datacentre using g729. The asterisk box in the datacentre connects to our other asterisk boxes providing pstn connectivity via IAX2. Latency between these boxes is between 1 and 2ms. The ADSL connections to the client sites are all consistently delivering latencies of sub-25ms to the datacentre and there is traffic shaping on that connection to give priority to any traffic from the phones' IPs. Comments from the users at these sites are as follows: call sounded like a dalek and I couldn't make out anything the caller was saying the phone on my desk is breaking up so badly it's virtually unusable calls sound like they're breaking up with metallic background noises We have quite a few customers with asterisk boxes on-site (with phones connected to them via the LAN) using ADSL connections from the same supplier, and are not having these issues with them. canreinvite=no and nat=yes are set on all these devices, since they are behind NAT. Each device re-registers with asterisk every 5 minutes to prevent any possible NAT state timeouts. Any pointers/places to look for potential problems would be much appreciated. Regards, Chris ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Diagnosing poor call quality
Eric said: This should be a FAQ. Set the RTP packet size on the SPAs to .2 instead of .3 Thanks for the suggestion. I've logged into the offending devices and set both to .2. I'll see how it goes for 48 hours or so. I've looked at the Elmeg ip290's and they are set to 20ms from factory, so I don't think that's the issue there. Michael said: The advertised datarate (8mb/448k) are the speeds at which the circuit between the customer and the central office is clocked and has no relationship with *effective* throughput. I have run a few speed tests from the sites in question (iperf to the machine in the datacentre) and I'm consistently getting around 380k upstream and 5.5mbit downstream, even during peak hours. Some distance away from the quoted speeds, but still plenty enough to support 4 SIP devices using g729 (which should be about 30kbit/sec per device including packet overheads). So beware of VoIP over DSL. Your users need to be aware of the tradeoffs between the cost of DSL vs. T1 and the effect on call quality. Alas, T1 for net traffic here in the UK is insanely expensive. DSL in its various forms is about the best we get, and SDSL with low contention ratios (1:1, 5:1, etc.) is only available in a few exchanges in major cities. Chris, if your customers are in the western US then please contact me about dedicated circuits. About 4500 miles away. :-) Thanks for the offer anyway though. Any further thoughts would be gratefully appreciated, especially for the site with the Elmegs. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited This email is made from 100% recycled electrons ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
