[asterisk-users] bad call quality
Hello Im at a complete loss. I run a couple of asterisk servers all connecting to international sip providers. All three servers are on the same type of internet connection (Martis/Diginet). There isnt a shortage of bandwidth, and its not a codec issue, as ive tried swapping codecs. If its not a line issue, because if i route the calls via sip via another server(which i own)(in same country) and then break out from there i get good quality, but im paying for triple bandwidth then, and bandwidth in south Africa is hellishly expensive. The Physical hardware is not overloaded either. I have tried rebooting my equipment, and that changed nothing either. if i do a ping flood i get decent results(well, only about 10ms more than another perfectly working branch) What else could this Be? Im completely Dumbstruck. Regards Edd ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] bad call quality
Hi Edd - I run a couple of asterisk servers all connecting to international sip providers. All three servers are on the same type of internet connection (Martis/Diginet). There isnt a shortage of bandwidth, and its not a codec issue, as ive tried swapping codecs. If its not a line issue, because if i route the calls via sip via another server(which i own)(in same country) and then break out from there i get good quality, but im paying for triple bandwidth then, and bandwidth in south Africa is hellishly expensive. The Physical hardware is not overloaded either. I have tried rebooting my equipment, and that changed nothing either. if i do a ping flood i get decent results(well, only about 10ms more than another perfectly working branch) What else could this Be? Im completely Dumbstruck. Is there any other non-VoIP traffic using the same internet connection as the asterisk server? If so, this could very well be a QoS issue. You can get some nasty sounding calls even on a very fat internet connection if there is no QoS. One of my clients has a 100mb fiber connection to the internet, and we had to really fine tune their Cisco routers in order to get usable VoIP calls to their branch offices. I've also seen internet connections that are just very poor, and no amount of internal QoS can fix this. What kind of routing equipment are you using? - Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] bad call quality
Noah Miller wrote: Hi Edd - I run a couple of asterisk servers all connecting to international sip providers. All three servers are on the same type of internet connection (Martis/Diginet). There isnt a shortage of bandwidth, and its not a codec issue, as ive tried swapping codecs. If its not a line issue, because if i route the calls via sip via another server(which i own)(in same country) and then break out from there i get good quality, but im paying for triple bandwidth then, and bandwidth in south Africa is hellishly expensive. The Physical hardware is not overloaded either. I have tried rebooting my equipment, and that changed nothing either. if i do a ping flood i get decent results(well, only about 10ms more than another perfectly working branch) What else could this Be? Im completely Dumbstruck. Is there any other non-VoIP traffic using the same internet connection as the asterisk server? If so, this could very well be a QoS issue. You can get some nasty sounding calls even on a very fat internet connection if there is no QoS. One of my clients has a 100mb fiber connection to the internet, and we had to really fine tune their Cisco routers in order to get usable VoIP calls to their branch offices. I've also seen internet connections that are just very poor, and no amount of internal QoS can fix this. What kind of routing equipment are you using? - Noah Im Using a cisco, but the internet connection is dedicated to VOIP. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Bad call quality using a certain channel.
Hi, I am fairly new at working with Asterisk. I am having a call quality issue that I really need to get ironed out before we go to rollout the system in a week. Any help would be greatly appreciated!!! Even if it is just pointing me in the right direction. My current setup: I have Asterisk Setup on a Dell Server. It has 2 T100P cards. One will be for out T1 PRI from the Phone Company (We dont have this installed yet) The other T100P connects to a VINA T1 IAD (Channel Bank) I also have a Cisco 7960 SIP Phone attached and registered. The Server is connected to a broadband connection. My issue: When I call the IAX Demo from the SIP Phone, the call is perfect. Asterisk Voice is 100%, and the Voice from the Digium Test server is almost 100% (an occasional stutter)..but very usable. When I call the IAX Demo from a Phone connected to the VINA Channel Bank, the Asterisk Voice is 100%, but once it connects to the test server it is extremely choppy. You can kind of understand what is being said, but it is very very poor quality and quite unusable. When I between the Channel Bank and the SIP phone, the quality is 100% no problems at all. So..why does the VINA Channel Bank connection not seem to like the IAX side of things, When I know that the IAX side is functioning great when used from a SIP Phone? I dont know what details would be pertinent to this, but here is what the Asterisk Console Displays: -- Executing Playback(SIP/200-ad26, demo-abouttotry) in new stack -- Playing 'demo-abouttotry' (language 'en') -- Executing Dial(SIP/200-ad26, IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED]/[EMAIL PROTECTED] -- Call accepted by 216.207.245.8 (format gsm) -- Format for call is gsm -- IAX2/216.207.245.8:4569-1 is ringing -- IAX2/216.207.245.8:4569-1 answered SIP/200-ad26 -- Hungup 'IAX2/216.207.245.8:4569-1' == Spawn extension (from-sip, 861, 2) exited non-zero on 'SIP/200-ad26' asterisk1*CLI asterisk1*CLI -- Starting simple switch on 'Zap/25-1' -- Executing Playback(Zap/25-1, demo-abouttotry) in new stack -- Playing 'demo-abouttotry' (language 'en') -- Executing Dial(Zap/25-1, IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED]/[EMAIL PROTECTED] -- Call accepted by 216.207.245.8 (format gsm) -- Format for call is gsm -- IAX2/216.207.245.8:4569-2 is ringing -- IAX2/216.207.245.8:4569-2 answered Zap/25-1 -- THIS IS WHERE THE AUDIO BECOMES ALL CHOPPED UP. -- Hungup 'IAX2/216.207.245.8:4569-2' == Spawn extension (chan_bank, 861, 2) exited non-zero on 'Zap/25-1' -- Hungup 'Zap/25-1' asterisk1*CLI ___ --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
