Re: [Asterisk-Users] GXP-2000 Audio Quality
Tim Panton wrote: Well, with 16 phones, it might be worth putting a 'satellite' asterisk in their office, have it handle local transfers, and act as a protocol converter, talking sip to the phones and (trunked) IAX2 to the outside world. An embedded low power system would do fine. You might even get away with an nslu2, but I'm not sure it has the RAM for 16 calls. A better alternative is to get them to upgrade the DSL to 512 uplink. Tim. Neither the unslung nor the wrt support IAX trunking. Zaptel does not compile on either of these architectures. No zaptel = no timer = no trunking/meetme/etc. -- Kristian Kielhofner ___ --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] GXP-2000 Audio Quality
On 17 Jun 2006, at 07:53, Kristian Kielhofner wrote: Tim Panton wrote: Well, with 16 phones, it might be worth putting a 'satellite' asterisk in their office, have it handle local transfers, and act as a protocol converter, talking sip to the phones and (trunked) IAX2 to the outside world. An embedded low power system would do fine. You might even get away with an nslu2, but I'm not sure it has the RAM for 16 calls. A better alternative is to get them to upgrade the DSL to 512 uplink. Tim. Neither the unslung nor the wrt support IAX trunking. Zaptel does not compile on either of these architectures. No zaptel = no timer = no trunking/meetme/etc. Just out of curiosity, is ztdummy on kernel 2.6.12.2 architecture specific? i.e. would it care if it were on an armv5teb not on x86 ? I understand that the _real_ zaptel modules will be much harder to port, I just figured that ztdummy might be easier. Tim. Tim Panton [EMAIL PROTECTED] ___ --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] GXP-2000 Audio Quality
On Sat, Jun 17, 2006 at 11:14:33AM +0100, Tim Panton wrote: On 17 Jun 2006, at 07:53, Kristian Kielhofner wrote: Tim Panton wrote: Well, with 16 phones, it might be worth putting a 'satellite' asterisk in their office, have it handle local transfers, and act as a protocol converter, talking sip to the phones and (trunked) IAX2 to the outside world. An embedded low power system would do fine. You might even get away with an nslu2, but I'm not sure it has the RAM for 16 calls. A better alternative is to get them to upgrade the DSL to 512 uplink. Tim. Neither the unslung nor the wrt support IAX trunking. Zaptel does not compile on either of these architectures. No zaptel = no timer = no trunking/meetme/etc. Just out of curiosity, is ztdummy on kernel 2.6.12.2 architecture specific? i.e. would it care if it were on an armv5teb not on x86 ? ztdummy on kernel 2.6 has tw implementations: with USE_RTC defined (the default on x86, at least) it uses the rtc clock of the system. This is availble on x86 and amd64. I don't know if other architectures have anything equivalent. Without it, it relies on HZ=1000 . That was the only possible value up until 2.6.13 , so I guess that in the specific kernel you refer to it should hold. I understand that the _real_ zaptel modules will be much harder to port, I just figured that ztdummy might be easier. Most other modules are PCI cards. Two others are USB. I don't know how much architecture-specific are PCI and USB. There are also ztdynamic and friends. In theory nothing prevents them from being portable. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.com ___ --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] GXP-2000 Audio Quality
G729 uses 8kbps but with the IP overhead it actually uses 30kbps so for 256k upstream you should be able to handle 8 calls but this is in ideal conditions. If you were to use IAX and enable trunking then you would use 30kbps for the 1st call and 10kbps for each additional call. See http://www.voip-info.org/wiki/index.php?page=Asterisk+bandwidth+iax2 On Wed, 2006-06-14 at 04:17, Daniel Salama wrote: I have a client with about 16 GXP-2000. They complain that the audio quality is terrible after 2 or 3 simultaneous conversations. They are behind DSL 1.5Mbps down and 256Kbps up. Because they are using G711.u codec, I know they upstream bandwidth is the limiting factor and they most likely won't be able to have more than 3 simultaneous conversations, and if they're surfing the net and/or checking email, things will only get worse. So, I purchased some g729 codec licenses and forced their sip peer configuration to g729 codec. We made sample test calls and were able to make 8 simultaneous calls. On the eighth call, the audio started to sound choppy. Then we dropped the eighth call and tested with 7. We could hear just fine on the GXP-2000 but the remote end heard us a bit choppy and/or with a robot-like voice. So, we kept dropping calls until they were of acceptable quality. My question is, if they were using g729 which, in theory uses 8kbps plus overhead, they should have been just fine handling eight calls. All the computers were turned off on the network, so there shouldn't have been any other traffic but VoIP. Does anyone have any ideas? How can I improve their audio quality? I requested BellSouth to upgrade their capacity but because of where they are located, the best they can get is 900Kbps/256Kbps, so the upstream continues to be the limiting factor. I purchased a Dlink-1226G switch to allow me to control QoS on the LAN. I also upgraded their Netopia DSL router to the latest firmware which allows me to configure VLANs and DiffServ. All the computers are connected to the PC port on the phone because there is no available second wiring. Can anyone suggest how to configure the QoS settings on the phones, the Dlink and the Netopia? While there was no traffic on the wire, pinging from/to the Asterisk box gave me about 47ms latency. When we went passed the 4th call, the latency started increasing significantly and when we got to 8 calls, the latency was up in the 2000ms. Obviously, if anything I did in the QoS configuration gave VoIP a priority, then ICMP packets would have the lowest priority and I could understand that to be the reason for such result. However, I'm not sure I configured QoS properly and that's why I'm asking for help. Thanks, Daniel ___ --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] GXP-2000 Audio Quality
Wow! 22Kbps of overhead? Are you sure? That sounds like way too much overhead. I can't use IAX2 because the GXP-2000 are SIP phones :( Any other suggestion? Thanks, Daniel On Jun 14, 2006, at 4:37 AM, Gareth Blades wrote: G729 uses 8kbps but with the IP overhead it actually uses 30kbps so for 256k upstream you should be able to handle 8 calls but this is in ideal conditions. If you were to use IAX and enable trunking then you would use 30kbps for the 1st call and 10kbps for each additional call. See http://www.voip-info.org/wiki/index.php?page=Asterisk+bandwidth +iax2 On Wed, 2006-06-14 at 04:17, Daniel Salama wrote: I have a client with about 16 GXP-2000. They complain that the audio quality is terrible after 2 or 3 simultaneous conversations. They are behind DSL 1.5Mbps down and 256Kbps up. Because they are using G711.u codec, I know they upstream bandwidth is the limiting factor and they most likely won't be able to have more than 3 simultaneous conversations, and if they're surfing the net and/or checking email, things will only get worse. So, I purchased some g729 codec licenses and forced their sip peer configuration to g729 codec. We made sample test calls and were able to make 8 simultaneous calls. On the eighth call, the audio started to sound choppy. Then we dropped the eighth call and tested with 7. We could hear just fine on the GXP-2000 but the remote end heard us a bit choppy and/or with a robot-like voice. So, we kept dropping calls until they were of acceptable quality. My question is, if they were using g729 which, in theory uses 8kbps plus overhead, they should have been just fine handling eight calls. All the computers were turned off on the network, so there shouldn't have been any other traffic but VoIP. Does anyone have any ideas? How can I improve their audio quality? I requested BellSouth to upgrade their capacity but because of where they are located, the best they can get is 900Kbps/256Kbps, so the upstream continues to be the limiting factor. I purchased a Dlink-1226G switch to allow me to control QoS on the LAN. I also upgraded their Netopia DSL router to the latest firmware which allows me to configure VLANs and DiffServ. All the computers are connected to the PC port on the phone because there is no available second wiring. Can anyone suggest how to configure the QoS settings on the phones, the Dlink and the Netopia? While there was no traffic on the wire, pinging from/to the Asterisk box gave me about 47ms latency. When we went passed the 4th call, the latency started increasing significantly and when we got to 8 calls, the latency was up in the 2000ms. Obviously, if anything I did in the QoS configuration gave VoIP a priority, then ICMP packets would have the lowest priority and I could understand that to be the reason for such result. However, I'm not sure I configured QoS properly and that's why I'm asking for help. Thanks, Daniel ___ --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 ___ --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] GXP-2000 Audio Quality
Welcome to the wonderful world of VoIP, where people are eager to move from 8kbps G.729 to 6.3kbps G.723.1, and accept a substantial drop in voice quality, and then throw over 20kbps of RTP, IP and related overhead on top of them. Isn't IP wonderful? :-) Regards, Steve Daniel Salama wrote: Wow! 22Kbps of overhead? Are you sure? That sounds like way too much overhead. I can't use IAX2 because the GXP-2000 are SIP phones :( Any other suggestion? Thanks, Daniel On Jun 14, 2006, at 4:37 AM, Gareth Blades wrote: G729 uses 8kbps but with the IP overhead it actually uses 30kbps so for 256k upstream you should be able to handle 8 calls but this is in ideal conditions. If you were to use IAX and enable trunking then you would use 30kbps for the 1st call and 10kbps for each additional call. See http://www.voip-info.org/wiki/index.php?page=Asterisk+bandwidth +iax2 On Wed, 2006-06-14 at 04:17, Daniel Salama wrote: I have a client with about 16 GXP-2000. They complain that the audio quality is terrible after 2 or 3 simultaneous conversations. They are behind DSL 1.5Mbps down and 256Kbps up. Because they are using G711.u codec, I know they upstream bandwidth is the limiting factor and they most likely won't be able to have more than 3 simultaneous conversations, and if they're surfing the net and/or checking email, things will only get worse. So, I purchased some g729 codec licenses and forced their sip peer configuration to g729 codec. We made sample test calls and were able to make 8 simultaneous calls. On the eighth call, the audio started to sound choppy. Then we dropped the eighth call and tested with 7. We could hear just fine on the GXP-2000 but the remote end heard us a bit choppy and/or with a robot-like voice. So, we kept dropping calls until they were of acceptable quality. My question is, if they were using g729 which, in theory uses 8kbps plus overhead, they should have been just fine handling eight calls. All the computers were turned off on the network, so there shouldn't have been any other traffic but VoIP. Does anyone have any ideas? How can I improve their audio quality? I requested BellSouth to upgrade their capacity but because of where they are located, the best they can get is 900Kbps/256Kbps, so the upstream continues to be the limiting factor. I purchased a Dlink-1226G switch to allow me to control QoS on the LAN. I also upgraded their Netopia DSL router to the latest firmware which allows me to configure VLANs and DiffServ. All the computers are connected to the PC port on the phone because there is no available second wiring. Can anyone suggest how to configure the QoS settings on the phones, the Dlink and the Netopia? While there was no traffic on the wire, pinging from/to the Asterisk box gave me about 47ms latency. When we went passed the 4th call, the latency started increasing significantly and when we got to 8 calls, the latency was up in the 2000ms. Obviously, if anything I did in the QoS configuration gave VoIP a priority, then ICMP packets would have the lowest priority and I could understand that to be the reason for such result. However, I'm not sure I configured QoS properly and that's why I'm asking for help. Thanks, Daniel ___ --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] GXP-2000 Audio Quality
Well, with 16 phones, it might be worth putting a 'satellite' asterisk in their office, have it handle local transfers, and act as a protocol converter, talking sip to the phones and (trunked) IAX2 to the outside world. An embedded low power system would do fine. You might even get away with an nslu2, but I'm not sure it has the RAM for 16 calls. A better alternative is to get them to upgrade the DSL to 512 uplink. Tim. On 14 Jun 2006, at 17:11, Daniel Salama wrote: Wow! 22Kbps of overhead? Are you sure? That sounds like way too much overhead. I can't use IAX2 because the GXP-2000 are SIP phones :( Any other suggestion? Thanks, Daniel On Jun 14, 2006, at 4:37 AM, Gareth Blades wrote: G729 uses 8kbps but with the IP overhead it actually uses 30kbps so for 256k upstream you should be able to handle 8 calls but this is in ideal conditions. If you were to use IAX and enable trunking then you would use 30kbps for the 1st call and 10kbps for each additional call. See http://www.voip-info.org/wiki/index.php?page=Asterisk+bandwidth +iax2 On Wed, 2006-06-14 at 04:17, Daniel Salama wrote: I have a client with about 16 GXP-2000. They complain that the audio quality is terrible after 2 or 3 simultaneous conversations. They are behind DSL 1.5Mbps down and 256Kbps up. Because they are using G711.u codec, I know they upstream bandwidth is the limiting factor and they most likely won't be able to have more than 3 simultaneous conversations, and if they're surfing the net and/or checking email, things will only get worse. So, I purchased some g729 codec licenses and forced their sip peer configuration to g729 codec. We made sample test calls and were able to make 8 simultaneous calls. On the eighth call, the audio started to sound choppy. Then we dropped the eighth call and tested with 7. We could hear just fine on the GXP-2000 but the remote end heard us a bit choppy and/or with a robot-like voice. So, we kept dropping calls until they were of acceptable quality. My question is, if they were using g729 which, in theory uses 8kbps plus overhead, they should have been just fine handling eight calls. All the computers were turned off on the network, so there shouldn't have been any other traffic but VoIP. Does anyone have any ideas? How can I improve their audio quality? I requested BellSouth to upgrade their capacity but because of where they are located, the best they can get is 900Kbps/256Kbps, so the upstream continues to be the limiting factor. I purchased a Dlink-1226G switch to allow me to control QoS on the LAN. I also upgraded their Netopia DSL router to the latest firmware which allows me to configure VLANs and DiffServ. All the computers are connected to the PC port on the phone because there is no available second wiring. Can anyone suggest how to configure the QoS settings on the phones, the Dlink and the Netopia? While there was no traffic on the wire, pinging from/to the Asterisk box gave me about 47ms latency. When we went passed the 4th call, the latency started increasing significantly and when we got to 8 calls, the latency was up in the 2000ms. Obviously, if anything I did in the QoS configuration gave VoIP a priority, then ICMP packets would have the lowest priority and I could understand that to be the reason for such result. However, I'm not sure I configured QoS properly and that's why I'm asking for help. Thanks, Daniel ___ --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 ___ --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 Tim Panton [EMAIL PROTECTED] ___ --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] GXP-2000 Audio Quality
That may not be such a bad idea. I've read people trying to put Asterisk on a WRTG54 or something like that. Would that be good? I guess I could do SIP in the office and trunk via IAX2 and save on bandwidth plus internal calls would be local. I tried to upgrade them to 512K but because they're borderline to the 18K feet, the best BellSouth can offer them is 256K. I'm talking to Comcast to see if they can get their broadband service which can go up to 768K. Thanks, Daniel On Jun 14, 2006, at 12:45 PM, Tim Panton wrote: Well, with 16 phones, it might be worth putting a 'satellite' asterisk in their office, have it handle local transfers, and act as a protocol converter, talking sip to the phones and (trunked) IAX2 to the outside world. An embedded low power system would do fine. You might even get away with an nslu2, but I'm not sure it has the RAM for 16 calls. A better alternative is to get them to upgrade the DSL to 512 uplink. Tim. On 14 Jun 2006, at 17:11, Daniel Salama wrote: Wow! 22Kbps of overhead? Are you sure? That sounds like way too much overhead. I can't use IAX2 because the GXP-2000 are SIP phones :( Any other suggestion? Thanks, Daniel On Jun 14, 2006, at 4:37 AM, Gareth Blades wrote: G729 uses 8kbps but with the IP overhead it actually uses 30kbps so for 256k upstream you should be able to handle 8 calls but this is in ideal conditions. If you were to use IAX and enable trunking then you would use 30kbps for the 1st call and 10kbps for each additional call. See http://www.voip-info.org/wiki/index.php?page=Asterisk +bandwidth+iax2 On Wed, 2006-06-14 at 04:17, Daniel Salama wrote: I have a client with about 16 GXP-2000. They complain that the audio quality is terrible after 2 or 3 simultaneous conversations. They are behind DSL 1.5Mbps down and 256Kbps up. Because they are using G711.u codec, I know they upstream bandwidth is the limiting factor and they most likely won't be able to have more than 3 simultaneous conversations, and if they're surfing the net and/or checking email, things will only get worse. So, I purchased some g729 codec licenses and forced their sip peer configuration to g729 codec. We made sample test calls and were able to make 8 simultaneous calls. On the eighth call, the audio started to sound choppy. Then we dropped the eighth call and tested with 7. We could hear just fine on the GXP-2000 but the remote end heard us a bit choppy and/or with a robot-like voice. So, we kept dropping calls until they were of acceptable quality. My question is, if they were using g729 which, in theory uses 8kbps plus overhead, they should have been just fine handling eight calls. All the computers were turned off on the network, so there shouldn't have been any other traffic but VoIP. Does anyone have any ideas? How can I improve their audio quality? I requested BellSouth to upgrade their capacity but because of where they are located, the best they can get is 900Kbps/256Kbps, so the upstream continues to be the limiting factor. I purchased a Dlink-1226G switch to allow me to control QoS on the LAN. I also upgraded their Netopia DSL router to the latest firmware which allows me to configure VLANs and DiffServ. All the computers are connected to the PC port on the phone because there is no available second wiring. Can anyone suggest how to configure the QoS settings on the phones, the Dlink and the Netopia? While there was no traffic on the wire, pinging from/to the Asterisk box gave me about 47ms latency. When we went passed the 4th call, the latency started increasing significantly and when we got to 8 calls, the latency was up in the 2000ms. Obviously, if anything I did in the QoS configuration gave VoIP a priority, then ICMP packets would have the lowest priority and I could understand that to be the reason for such result. However, I'm not sure I configured QoS properly and that's why I'm asking for help. Thanks, Daniel ___ --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 ___ --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 Tim Panton [EMAIL PROTECTED] ___ --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] GXP-2000 Audio Quality
I have a client with about 16 GXP-2000. They complain that the audio quality is terrible after 2 or 3 simultaneous conversations. They are behind DSL 1.5Mbps down and 256Kbps up. Because they are using G711.u codec, I know they upstream bandwidth is the limiting factor and they most likely won't be able to have more than 3 simultaneous conversations, and if they're surfing the net and/or checking email, things will only get worse. So, I purchased some g729 codec licenses and forced their sip peer configuration to g729 codec. We made sample test calls and were able to make 8 simultaneous calls. On the eighth call, the audio started to sound choppy. Then we dropped the eighth call and tested with 7. We could hear just fine on the GXP-2000 but the remote end heard us a bit choppy and/or with a robot-like voice. So, we kept dropping calls until they were of acceptable quality. My question is, if they were using g729 which, in theory uses 8kbps plus overhead, they should have been just fine handling eight calls. All the computers were turned off on the network, so there shouldn't have been any other traffic but VoIP. Does anyone have any ideas? How can I improve their audio quality? I requested BellSouth to upgrade their capacity but because of where they are located, the best they can get is 900Kbps/256Kbps, so the upstream continues to be the limiting factor. I purchased a Dlink-1226G switch to allow me to control QoS on the LAN. I also upgraded their Netopia DSL router to the latest firmware which allows me to configure VLANs and DiffServ. All the computers are connected to the PC port on the phone because there is no available second wiring. Can anyone suggest how to configure the QoS settings on the phones, the Dlink and the Netopia? While there was no traffic on the wire, pinging from/to the Asterisk box gave me about 47ms latency. When we went passed the 4th call, the latency started increasing significantly and when we got to 8 calls, the latency was up in the 2000ms. Obviously, if anything I did in the QoS configuration gave VoIP a priority, then ICMP packets would have the lowest priority and I could understand that to be the reason for such result. However, I'm not sure I configured QoS properly and that's why I'm asking for help. Thanks, Daniel ___ --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
