Re: [Asterisk-Users] need help troubleshooting clipping and garbled VOIP calls
T. Shaw wrote: Hello all, I have a problem with call quality with my Asterisk setup. I'm doing VOIP only so far, but have a zaptel TDM400P in the box not being used. The problem i'm having is that when calls are placed, connected, and the far-end is reporting that they are experiencing clipping, choppy, and garbled voice conversations. So bad that we have to resort to using our cell phones. This entire setup is still being built, but any phone attached is experiencing this. Call volume is almost nil (under 20 total incoming calls a day). This is a small business setup. The server is used exclusively for Asterisk, so it isn't a fileserver, or anything else. The setup is as such: ipphone ---cisco 2900XL switch Cisco 2621 router --- dsl modem --DSL --- VOIPprovider I've configured the switch and the router to set priority and qos to prioritize voice traffic above data. Funny thing is, there is not data REALLY hitting the network. I have setup 2 vlans, data vlan, and voice vlan. There are two work stations on the network, and neither is being used to hit the internet heavily (office is still being setup). Any pointers or suggestions anyone have for me as to were to look for this poor quality? It seems only the Far-end (called party), is hearing this and not the calling party. I haven't tried switching out the phones because we only have 1 type, and any of the phones i used exhibit these problems. I will try softphones to see if it is truly a networking issue or Phone issue. Is anyone using a cisco 2900 switch or router and care to provide config samples of their COS/QOS setup? Thanks! Terrelle Shaw I've got a similar setup (which does have a TDM card and voip incoming and outgoing), for some reason an IAX provider (which provides most of our calls incoming and outgoing) has this problem, whereas a different SIP one doesn't seem to. I have checked my traffic shaping script, and everything seems fine, the same provider works flawlesly from home, with a simliar setup (only without a timing source and a cable modem). I'd be very interested to see what you find out. ___ --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] need help troubleshooting clipping and garbled VOIP calls
Martin Joseph wrote: On Jun 29, 2006, at 2:43 PM, T. Shaw wrote: thanks for all the responses. I feared that it might be a bandwidth issue. We have a (supposedly) business DSL line that is 1.5M - 3M down/ 512k up. might have to bump that up to a higher grade. If you are actually getting what you describe above you should have no need to upgrade in order to support 3 or 4 simultaneous calls... I have only 384K bit's upstream on my home DSL and that's fine for 2 uLaw calls. If you switch to GSM or G729 that should allow for even greater simultaneous call volume... Accorging to the calculator on asterisk-guru (which I know isn't perfect), you should be able to manage at least 30 calls with trunked IAX and G.729. I did take your suggestion and contact my VOIP provider. They suggested to two things: 1) Use SIP to trunk with them instead of IAX ( they said that lots of people complain about the conenction with IAX, but when they use SIP the issues get better) ? This sounds like they have an issue. With (presumably) a different provider, I seem to be getting a similar problem, at home (even with the same accounts) IAX-in and IAX-out causes no problems, with G.729, whereas on site IAX-in and IAX-out calls have clipping and buzzing, with same handsets, same codec only differences being that on site there is a timing source (TDM400) and the IAX channels are trunked. SIP-in doesn't seem to cause the same problems on-site. ___ --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
[Asterisk-Users] need help troubleshooting clipping and garbled VOIP calls
Hello all, I have a problem with call quality with my Asterisk setup. I'm doing VOIP only so far, but have a zaptel TDM400P in the box not being used. The problem i'm having is that when calls are placed, connected, and the far-end is reporting that they are experiencing clipping, choppy, and garbled voice conversations. So bad that we have to resort to using our cell phones. This entire setup is still being built, but any phone attached is experiencing this. Call volume is almost nil (under 20 total incoming calls a day). This is a small business setup. The server is used exclusively for Asterisk, so it isn't a fileserver, or anything else. The setup is as such: ipphone ---cisco 2900XL switch Cisco 2621 router --- dsl modem --DSL --- VOIPprovider I've configured the switch and the router to set priority and qos to prioritize voice traffic above data. Funny thing is, there is not data REALLY hitting the network. I have setup 2 vlans, data vlan, and voice vlan. There are two work stations on the network, and neither is being used to hit the internet heavily (office is still being setup). Any pointers or suggestions anyone have for me as to were to look for this poor quality? It seems only the Far-end (called party), is hearing this and not the calling party. I haven't tried switching out the phones because we only have 1 type, and any of the phones i used exhibit these problems. I will try softphones to see if it is truly a networking issue or Phone issue. Is anyone using a cisco 2900 switch or router and care to provide config samples of their COS/QOS setup? Thanks! Terrelle Shaw ___ --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] need help troubleshooting clipping and garbled VOIP calls
T. Shaw wrote: Hello all, I have a problem with call quality with my Asterisk setup. I'm doing VOIP only so far, but have a zaptel TDM400P in the box not being used. The problem i'm having is that when calls are placed, connected, and the far-end is reporting that they are experiencing clipping, choppy, and garbled voice conversations. So bad that we have to resort to using our cell phones. This entire setup is still being built, but any phone attached is experiencing this. Call volume is almost nil (under 20 total incoming calls a day). This is a small business setup. The server is used exclusively for Asterisk, so it isn't a fileserver, or anything else. The setup is as such: ipphone ---cisco 2900XL switch Cisco 2621 router --- dsl modem --DSL --- VOIPprovider I've configured the switch and the router to set priority and qos to prioritize voice traffic above data. Funny thing is, there is not data REALLY hitting the network. I have setup 2 vlans, data vlan, and voice vlan. There are two work stations on the network, and neither is being used to hit the internet heavily (office is still being setup). Any pointers or suggestions anyone have for me as to were to look for this poor quality? It seems only the Far-end (called party), is hearing this and not the calling party. I haven't tried switching out the phones because we only have 1 type, and any of the phones i used exhibit these problems. I will try softphones to see if it is truly a networking issue or Phone issue. Is anyone using a cisco 2900 switch or router and care to provide config samples of their COS/QOS setup? Highly unlikely the ipphones or switch have anything to do with the problem. Most likely cause is the dsl upload speed (which is usually substantially slower then download speed). Just as an FYI... the QoS settings in any Cisco product (including the 2900 switch) do not actually kick in until a switch port becomes congested. So if your switch ports are running at only 10 megabits, you'd have to be moving data (including voice) at that rate before QoS begins to give priority to voice packets. If your voip provider supports a low bit-rate codec, you might consider using something like g729 to see if that impacts the problem. If it does improve it, then either your dsl connection is the problem (to slow), or, your dsl provider's data network is less then adequate to move the voice packets through in a reliable way. There are several internet sites that you can use to test your bandwidth if you want to try those. One is dslreports.com, but there are others. Goggle should give you plenty to work with. ___ --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] need help troubleshooting clipping and garbled VOIP calls
thanks for all the responses. I feared that it might be a bandwidth issue. We have a (supposedly) business DSL line that is 1.5M - 3M down/ 512k up. might have to bump that up to a higher grade. I did take your suggestion and contact my VOIP provider. They suggested to two things: 1) Use SIP to trunk with them instead of IAX ( they said that lots of people complain about the conenction with IAX, but when they use SIP the issues get better) 2) they can use GSM, and to try that as well (they also use Asterisk on their end) I first tried switching to GSM to see if that clears up some quality issues ( since it was the quicker of the two). I'm doing some test calls now. Thanks again everyone who responded.. Terrelle From: Colin Anderson [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: 'Asterisk Users Mailing List - Non-Commercial Discussion'asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] need help troubleshooting clipping and garbled VOIP calls Date: Thu, 29 Jun 2006 14:33:21 -0600 It seems only the Far-end (called party), is hearing this and not the calling party. So, looks like the A in ADSL is showing here. What is your upstream bandwidth? If it is 1mbit this is the most likely cause. If your VoIP provider allows it, change your codec to GSM, which IMO has the best bandwidth management of the non-licensed codecs. It can only help. good luck, hth ___ --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 ___ --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] need help troubleshooting clipping and garbled VOIP calls
On Jun 29, 2006, at 2:43 PM, T. Shaw wrote: thanks for all the responses. I feared that it might be a bandwidth issue. We have a (supposedly) business DSL line that is 1.5M - 3M down/ 512k up. might have to bump that up to a higher grade. If you are actually getting what you describe above you should have no need to upgrade in order to support 3 or 4 simultaneous calls... I have only 384K bit's upstream on my home DSL and that's fine for 2 uLaw calls. If you switch to GSM or G729 that should allow for even greater simultaneous call volume... I did take your suggestion and contact my VOIP provider. They suggested to two things: 1) Use SIP to trunk with them instead of IAX ( they said that lots of people complain about the conenction with IAX, but when they use SIP the issues get better) ? This sounds like they have an issue. snip ___ --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
