Re: [asterisk-users] TCP port, VPN and resolving the cutting voice problem
What you probably have is a DSL MODEM that can act as a ROUTER but most likely doesn't have to. Your device probably has the same capabilities as most modems, the added features of NAT, DHCP, and whatever else. Normally you can disable that additional functionality. Now you just have a DSL modem. If you can turn off the ROUTER functions on the MODEM then you can use a Vyatta server to be a ROUTER that just so happens to be connected to DSL, but could just as easily be connected to a gigabit connection. Have you tried dumping IAX and using SIP? Have you verified that your bandwidth is saturated? Have you run NTOP or a similar tool to see what is eating all the bandwidth? I would start with the above because you have no idea what the problem is at this point. You need to come to a consensus of how many simultaneous calls are going to be allowed. You can QoS your VoIP all day long, but if one too many people get on the phone, everyone suffers. Once you get that number, you have to do the math as far as bandwidth to reserve and limit the calls on the Asterisk side. If this leaves you with less than enough bandwidth for business activities, you have to get more bandwidth, it is that simple. 1. No, I don't think so. Why do you? You want voice to be #1 correct? I presume your LAN connection is faster than your DSL. Any modern server can handle these chores. You are talking DSL, so I cannot imagine you have much call volume, setups and tear downs. Any G729 or codec conversion should be very light. If you are using G729 then set the phones to use it as well. You could probably run World Community Grid and consume all of your cycles without a hitch (not recommended, I use it for burn in on new machines) 2. Yes, you could setup a failover but I have servers with years of uptime and over a year of Asterisk not being restarted 1.0 and 1.2. Besides internal communication, would you not lose phone service now if your DSL ROUTER had to be rebooted? You don't need to activate the firewall if you feel NAT is adequate protection. QoS is your goal, the rest is just icing on the cake. 3. You are not tagging the packets for the ISP, you are controlling the rate at which protocols can consume on outbound traffic. You assign a port a piece of the pie, you have to let Vyatta know how big the pie is and how much of a slice each protocol gets. Inbound is a little trickier, what kind of DSL do you have, inbound may not be the problem. If it is, last I knew Vyatta used Rate-limiting which would essentially drop packets from the sender causing them to slow down, the protocols that you do not limit will not drop packets. http://en.wikipedia.org/wiki/Rate_limiting It has been a while since I looked at the latest and greatest or talked to the dev guys at Vyatta but they were discussing another method on the inbound side. Nevertheless, rate-limiting works for VoIP when correctly applied. Use google for God's sake. There are very well done videos and diagrams that are specific to Asterisk, Vyatta, and all of your questions. http://www.google.com/search?q=vyatta+asterisk+qos Thanks, Steve T On Sun, Dec 5, 2010 at 1:36 PM, bilal ghayyad bilmar...@yahoo.com wrote: Dear Steve; I am fully thanks for your advise and kindly help. I am asking about the ability to use vyatte hardware DSL router because of the following reasons: 1) I am afraid to make Asterisk the gateway for the whole network and this might effect on the performance and might cause a big load, u do not think so? 2) If any problem happened regarding to the QoS rules or regarding to the firewall or any other thing and they decided to do hardware restart for the server (or the PC machine), then the Asterisk will be restarted and that will effect on the telephony service at the site? 3) I am afraid if we applied the QoS and bandwidth divsion at Vyatte, and then we route the traffic to the DSL router (which will do the NAT to ISP), then all the QoS rules will be ignored (or become not effected)? What do u think? Again, special thanks for the guide and special help. Regards Bilal - I wouldn't bother with their hardware. You can run it on most servers providing the drivers for the hardware are supported. Just install it on a box with two NICs and put it between the router and your LAN, both static IPs, simple If I were you, I would find out what kind of DSL modem you have, but if it is doing NAT, DHCP, and all of that, you may be able to turn off everything except for the modem and use Vyatta for everything from NAT, DHCP, QoS, Squid, Firewall. In this case, one NIC would have your public IP, I suspect you would get it via DHCP or worst case, from your ISP, the second NIC is for the LAN, you can add more NICs for various purposes as well. I run Asterisk on Vyatta systems and it works great. No NAT issues with remote phones, QoS, and whatever else your
Re: [asterisk-users] TCP port, VPN and resolving the cutting voice problem
Dear Steve; Really until now, I am not able to know if Vyatta has a DSL router (hardware) that can be used to do the QoS and bandwidth management without need to download the software of Vyatte and install at the server? I am trying actually not to let all the traffic passing Asterisk server (where Vyatte is installed), because making asterisk to be the bottle neck, then it is not a reliable solution for the network. Does not think so? The DSL bandwidth is 1 Mbps, so it is not enough. The used codecs are G729 I am doing a ping, and no request time out .. but voice is cutting when other is browsing and downloading .. even no request time out ... but if others are not using internet for data browsing and downloading, voice is fine. And yes, I tried to use SIP instead of IAX, but also there is a problem in the voice when other are using the internet. What do u think? Regards Bilal --- On Mon, 12/6/10, Steve Totaro stot...@asteriskhelpdesk.com wrote: From: Steve Totaro stot...@asteriskhelpdesk.com Subject: Re: TCP port, VPN and resolving the cutting voice problem To: bilal ghayyad bilmar...@yahoo.com Cc: asterisk-users@lists.digium.com, eng_mohd_ta...@hotmail.com Date: Monday, December 6, 2010, 3:21 PM What you probably have is a DSL MODEM that can act as a ROUTER but most likely doesn't have to. Your device probably has the same capabilities as most modems, the added features of NAT, DHCP, and whatever else. Normally you can disable that additional functionality. Now you just have a DSL modem. If you can turn off the ROUTER functions on the MODEM then you can use a Vyatta server to be a ROUTER that just so happens to be connected to DSL, but could just as easily be connected to a gigabit connection. Have you tried dumping IAX and using SIP? Have you verified that your bandwidth is saturated? Have you run NTOP or a similar tool to see what is eating all the bandwidth? I would start with the above because you have no idea what the problem is at this point. You need to come to a consensus of how many simultaneous calls are going to be allowed. You can QoS your VoIP all day long, but if one too many people get on the phone, everyone suffers. Once you get that number, you have to do the math as far as bandwidth to reserve and limit the calls on the Asterisk side. If this leaves you with less than enough bandwidth for business activities, you have to get more bandwidth, it is that simple. 1. No, I don't think so. Why do you? You want voice to be #1 correct? I presume your LAN connection is faster than your DSL. Any modern server can handle these chores. You are talking DSL, so I cannot imagine you have much call volume, setups and tear downs. Any G729 or codec conversion should be very light. If you are using G729 then set the phones to use it as well. You could probably run World Community Grid and consume all of your cycles without a hitch (not recommended, I use it for burn in on new machines) 2. Yes, you could setup a failover but I have servers with years of uptime and over a year of Asterisk not being restarted 1.0 and 1.2. Besides internal communication, would you not lose phone service now if your DSL ROUTER had to be rebooted? You don't need to activate the firewall if you feel NAT is adequate protection. QoS is your goal, the rest is just icing on the cake. 3. You are not tagging the packets for the ISP, you are controlling the rate at which protocols can consume on outbound traffic. You assign a port a piece of the pie, you have to let Vyatta know how big the pie is and how much of a slice each protocol gets. Inbound is a little trickier, what kind of DSL do you have, inbound may not be the problem. If it is, last I knew Vyatta used Rate-limiting which would essentially drop packets from the sender causing them to slow down, the protocols that you do not limit will not drop packets. http://en.wikipedia.org/wiki/Rate_limiting It has been a while since I looked at the latest and greatest or talked to the dev guys at Vyatta but they were discussing another method on the inbound side. Nevertheless, rate-limiting works for VoIP when correctly applied. Use google for God's sake. There are very well done videos and diagrams that are specific to Asterisk, Vyatta, and all of your questions. http://www.google.com/search?q=vyatta+asterisk+qos Thanks, Steve T On Sun, Dec 5, 2010 at 1:36 PM, bilal ghayyad bilmar...@yahoo.com wrote: Dear Steve; I am fully thanks for your advise and kindly help. I am asking about the ability to use vyatte hardware DSL router because of the following reasons: 1) I am afraid to make Asterisk the gateway for the whole network and this might effect on the performance and might cause a big load, u do not think so? 2) If any problem happened regarding to the QoS rules or regarding to the firewall or any other thing and they decided to do hardware restart for the server (or the PC
Re: [asterisk-users] TCP port, VPN and resolving the cutting voice problem
If I were you I would visit their site! I seriously doubt that they have a DSL router. They came out with appliances, maybe they do. Go empower yourself and look at their offerings. The first thing they put out as an appliance was a Dell R200. That was cool because we used Dell R200s in our fly-away kits, two for redundancy, so instead of buying their marked up R200s we just loaded up our own. Not to be rude, but it is still business hours there, give them a call. I cannot spoon feed you anymore. I don't see how it could be a bottleneck, it is impossible, your DSL is the bottleneck. Unless you are using really old junk, and even then, 10BaseT would probably be sufficient. I also told you that you didn't have to put Asterisk on the Vyatta box, it is just something I do. I have not had a problem with 100meg links or really latent and slow VSAT links. It just works. I have no more answers for you since you are unwilling to try to answer them yourself first. Thanks, Steve T On Mon, Dec 6, 2010 at 4:10 PM, bilal ghayyad bilmar...@yahoo.com wrote: Dear Steve; Really until now, I am not able to know if Vyatta has a DSL router (hardware) that can be used to do the QoS and bandwidth management without need to download the software of Vyatte and install at the server? I am trying actually not to let all the traffic passing Asterisk server (where Vyatte is installed), because making asterisk to be the bottle neck, then it is not a reliable solution for the network. Does not think so? The DSL bandwidth is 1 Mbps, so it is not enough. The used codecs are G729 I am doing a ping, and no request time out .. but voice is cutting when other is browsing and downloading .. even no request time out ... but if others are not using internet for data browsing and downloading, voice is fine. And yes, I tried to use SIP instead of IAX, but also there is a problem in the voice when other are using the internet. What do u think? Regards Bilal --- On *Mon, 12/6/10, Steve Totaro stot...@asteriskhelpdesk.com* wrote: From: Steve Totaro stot...@asteriskhelpdesk.com Subject: Re: TCP port, VPN and resolving the cutting voice problem To: bilal ghayyad bilmar...@yahoo.com Cc: asterisk-users@lists.digium.com, eng_mohd_ta...@hotmail.com Date: Monday, December 6, 2010, 3:21 PM What you probably have is a DSL MODEM that can act as a ROUTER but most likely doesn't have to. Your device probably has the same capabilities as most modems, the added features of NAT, DHCP, and whatever else. Normally you can disable that additional functionality. Now you just have a DSL modem. If you can turn off the ROUTER functions on the MODEM then you can use a Vyatta server to be a ROUTER that just so happens to be connected to DSL, but could just as easily be connected to a gigabit connection. Have you tried dumping IAX and using SIP? Have you verified that your bandwidth is saturated? Have you run NTOP or a similar tool to see what is eating all the bandwidth? I would start with the above because you have no idea what the problem is at this point. You need to come to a consensus of how many simultaneous calls are going to be allowed. You can QoS your VoIP all day long, but if one too many people get on the phone, everyone suffers. Once you get that number, you have to do the math as far as bandwidth to reserve and limit the calls on the Asterisk side. If this leaves you with less than enough bandwidth for business activities, you have to get more bandwidth, it is that simple. 1. No, I don't think so. Why do you? You want voice to be #1 correct? I presume your LAN connection is faster than your DSL. Any modern server can handle these chores. You are talking DSL, so I cannot imagine you have much call volume, setups and tear downs. Any G729 or codec conversion should be very light. If you are using G729 then set the phones to use it as well. You could probably run World Community Grid and consume all of your cycles without a hitch (not recommended, I use it for burn in on new machines) 2. Yes, you could setup a failover but I have servers with years of uptime and over a year of Asterisk not being restarted 1.0 and 1.2. Besides internal communication, would you not lose phone service now if your DSL ROUTER had to be rebooted? You don't need to activate the firewall if you feel NAT is adequate protection. QoS is your goal, the rest is just icing on the cake. 3. You are not tagging the packets for the ISP, you are controlling the rate at which protocols can consume on outbound traffic. You assign a port a piece of the pie, you have to let Vyatta know how big the pie is and how much of a slice each protocol gets. Inbound is a little trickier, what kind of DSL do you have, inbound may not be the problem. If it is, last I knew Vyatta used Rate-limiting which would essentially drop packets from the sender causing them to slow down, the protocols that
Re: [asterisk-users] TCP port, VPN and resolving the cutting voice problem
I wouldn't bother with their hardware. You can run it on most servers providing the drivers for the hardware are supported. Just install it on a box with two NICs and put it between the router and your LAN, both static IPs, simple If I were you, I would find out what kind of DSL modem you have, but if it is doing NAT, DHCP, and all of that, you may be able to turn off everything except for the modem and use Vyatta for everything from NAT, DHCP, QoS, Squid, Firewall. In this case, one NIC would have your public IP, I suspect you would get it via DHCP or worst case, from your ISP, the second NIC is for the LAN, you can add more NICs for various purposes as well. I run Asterisk on Vyatta systems and it works great. No NAT issues with remote phones, QoS, and whatever else your imagination can come up with. I also install Webmin and NTOP. Just be aware that as soon as you activate the firewall, everything is blocked, so if you are going to use it as a firewall, get as many rules in place as you can think of. Thanks, Steve T On Thu, Dec 2, 2010 at 3:14 PM, bilal ghayyad bilmar...@yahoo.com wrote: Dear; I understood that Vyatta is the solution for the QoS, but I am not able to know if I can use a Vyatta hardware router to be DSL router and I set my QoS in it to resolve the voice problem. Is it possible? Thanks for the help. Regards Bilal Thanks all for ur participation and kindly advise. As I noticed that jitterbuffer could help if the ping does not have request time out but the voice is also cutting .. but in that case, I have to set the jitterbuffer at the IP Phones and Asterisk boxes. I have a polycom phone for example, and to set the jitterbuffer there are the following paramters: Payload Size Jitter Buffer Minimum Jitter Buffer Shrink Jitter Buffer Maximum When it use the minimum, and when it use the Shrink and when it use the maximum? If to look at the asterisk (in the SIP or IAX files) then there are a paramters for the jitterbuffer also, but really I am not able to know when to use this and when to use this: jenable, jbforce, jbmaxsize, jbresyncthreashold, jbimpl, jblog How to use the jbresyncthreashold? In which case? Regarding to the QoS, which will be need in case having a packet loose, correct? I just need to ask about something: What I will be able to do if my ISP did not setup the QoS at his side? What kind of settings I can do in my DSL router (in case of Cisco, or in case of Linksys that running linux firmware)? From the other side, if I used linux server to set the QoS, so do I have to let all the network elements to pass this linux server (so it will be the default gateway for other elements)? Appreciate the kindly help. Regards Bilal If getting a second circuit is out of the question. 1. Switch to SIP 2. Install and Learn Vyatta for QoS (Squid may help you quite a bit as well) as your router (or whatever you prefer) I use the paid versions of Vyatta but the free edition should be sufficient. I did the same setup over OpenVPN VSAT links in Iraq, 700ms ping times. I used GSM and some tricks on the Vyatta box. Originally, before I deployed the above, it was a wild west situation like what you have now. Going from G729 to GSM made a big improvement in conjunction with QoS. My theory on that is that G729 is already a very lossy codec, so any more loss, garbled audio. GSM is less lossy. Switch from IAX to SIP was another huge improvement, and then finally putting Vyatta and QoS as my router made calls almost crystal clear. There was the obvious lag time but users get used to that and wait a second or two before speaking so they don't talk over each other and the quality was five by five, except for solar flares, sandstorms, rain. Things beyond my control. Thanks, Steve T -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TCP port, VPN and resolving the cutting voice problem
Dear Steve; I am fully thanks for your advise and kindly help. I am asking about the ability to use vyatte hardware DSL router because of the following reasons: 1) I am afraid to make Asterisk the gateway for the whole network and this might effect on the performance and might cause a big load, u do not think so? 2) If any problem happened regarding to the QoS rules or regarding to the firewall or any other thing and they decided to do hardware restart for the server (or the PC machine), then the Asterisk will be restarted and that will effect on the telephony service at the site? 3) I am afraid if we applied the QoS and bandwidth divsion at Vyatte, and then we route the traffic to the DSL router (which will do the NAT to ISP), then all the QoS rules will be ignored (or become not effected)? What do u think? Again, special thanks for the guide and special help. Regards Bilal - I wouldn't bother with their hardware. You can run it on most servers providing the drivers for the hardware are supported. Just install it on a box with two NICs and put it between the router and your LAN, both static IPs, simple If I were you, I would find out what kind of DSL modem you have, but if it is doing NAT, DHCP, and all of that, you may be able to turn off everything except for the modem and use Vyatta for everything from NAT, DHCP, QoS, Squid, Firewall. In this case, one NIC would have your public IP, I suspect you would get it via DHCP or worst case, from your ISP, the second NIC is for the LAN, you can add more NICs for various purposes as well. I run Asterisk on Vyatta systems and it works great. No NAT issues with remote phones, QoS, and whatever else your imagination can come up with. I also install Webmin and NTOP. Just be aware that as soon as you activate the firewall, everything is blocked, so if you are going to use it as a firewall, get as many rules in place as you can think of. Thanks, Steve T On Thu, Dec 2, 2010 at 3:14 PM, bilal ghayyad bilmar...@yahoo.com wrote: Dear; I understood that Vyatta is the solution for the QoS, but I am not able to know if I can use a Vyatta hardware router to be DSL router and I set my QoS in it to resolve the voice problem. Is it possible? Thanks for the help. Regards Bilal Thanks all for ur participation and kindly advise. As I noticed that jitterbuffer could help if the ping does not have request time out but the voice is also cutting .. but in that case, I have to set the jitterbuffer at the IP Phones and Asterisk boxes. I have a polycom phone for example, and to set the jitterbuffer there are the following paramters: Payload Size Jitter Buffer Minimum Jitter Buffer Shrink Jitter Buffer Maximum When it use the minimum, and when it use the Shrink and when it use the maximum? If to look at the asterisk (in the SIP or IAX files) then there are a paramters for the jitterbuffer also, but really I am not able to know when to use this and when to use this: jenable, jbforce, jbmaxsize, jbresyncthreashold, jbimpl, jblog How to use the jbresyncthreashold? In which case? Regarding to the QoS, which will be need in case having a packet loose, correct? I just need to ask about something: What I will be able to do if my ISP did not setup the QoS at his side? What kind of settings I can do in my DSL router (in case of Cisco, or in case of Linksys that running linux firmware)? From the other side, if I used linux server to set the QoS, so do I have to let all the network elements to pass this linux server (so it will be the default gateway for other elements)? Appreciate the kindly help. Regards Bilal If getting a second circuit is out of the question. 1. Switch to SIP 2. Install and Learn Vyatta for QoS (Squid may help you quite a bit as well) as your router (or whatever you prefer) I use the paid versions of Vyatta but the free edition should be sufficient. I did the same setup over OpenVPN VSAT links in Iraq, 700ms ping times. I used GSM and some tricks on the Vyatta box. Originally, before I deployed the above, it was a wild west situation like what you have now. Going from G729 to GSM made a big improvement in conjunction with QoS. My theory on that is that G729 is already a very lossy codec, so any more loss, garbled audio. GSM is less lossy. Switch from IAX to SIP was another huge improvement, and then finally putting Vyatta and QoS as my router made calls almost crystal clear. There was the obvious lag time but users get used to that and wait a second or two before speaking so they don't talk over each other and the quality was five by five,
Re: [asterisk-users] TCP port, VPN and resolving the cutting voice problem
Thanks all for ur participation and kindly advise. As I noticed that jitterbuffer could help if the ping does not have request time out but the voice is also cutting .. but in that case, I have to set the jitterbuffer at the IP Phones and Asterisk boxes. I have a polycom phone for example, and to set the jitterbuffer there are the following paramters: Payload Size Jitter Buffer Minimum Jitter Buffer Shrink Jitter Buffer Maximum When it use the minimum, and when it use the Shrink and when it use the maximum? If to look at the asterisk (in the SIP or IAX files) then there are a paramters for the jitterbuffer also, but really I am not able to know when to use this and when to use this: jenable, jbforce, jbmaxsize, jbresyncthreashold, jbimpl, jblog How to use the jbresyncthreashold? In which case? Regarding to the QoS, which will be need in case having a packet loose, correct? I just need to ask about something: What I will be able to do if my ISP did not setup the QoS at his side? What kind of settings I can do in my DSL router (in case of Cisco, or in case of Linksys that running linux firmware)? From the other side, if I used linux server to set the QoS, so do I have to let all the network elements to pass this linux server (so it will be the default gateway for other elements)? Appreciate the kindly help. Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TCP port, VPN and resolving the cutting voice problem
On Thu, Dec 2, 2010 at 4:15 AM, bilal ghayyad bilmar...@yahoo.com wrote: Thanks all for ur participation and kindly advise. As I noticed that jitterbuffer could help if the ping does not have request time out but the voice is also cutting .. but in that case, I have to set the jitterbuffer at the IP Phones and Asterisk boxes. I have a polycom phone for example, and to set the jitterbuffer there are the following paramters: Payload Size Jitter Buffer Minimum Jitter Buffer Shrink Jitter Buffer Maximum When it use the minimum, and when it use the Shrink and when it use the maximum? If to look at the asterisk (in the SIP or IAX files) then there are a paramters for the jitterbuffer also, but really I am not able to know when to use this and when to use this: jenable, jbforce, jbmaxsize, jbresyncthreashold, jbimpl, jblog How to use the jbresyncthreashold? In which case? Regarding to the QoS, which will be need in case having a packet loose, correct? I just need to ask about something: What I will be able to do if my ISP did not setup the QoS at his side? What kind of settings I can do in my DSL router (in case of Cisco, or in case of Linksys that running linux firmware)? From the other side, if I used linux server to set the QoS, so do I have to let all the network elements to pass this linux server (so it will be the default gateway for other elements)? Appreciate the kindly help. Regards Bilal If getting a second circuit is out of the question. 1. Switch to SIP 2. Install and Learn Vyatta for QoS (Squid may help you quite a bit as well) as your router (or whatever you prefer) I use the paid versions of Vyatta but the free edition should be sufficient. I did the same setup over OpenVPN VSAT links in Iraq, 700ms ping times. I used GSM and some tricks on the Vyatta box. Originally, before I deployed the above, it was a wild west situation like what you have now. Going from G729 to GSM made a big improvement in conjunction with QoS. My theory on that is that G729 is already a very lossy codec, so any more loss, garbled audio. GSM is less lossy. Switch from IAX to SIP was another huge improvement, and then finally putting Vyatta and QoS as my router made calls almost crystal clear. There was the obvious lag time but users get used to that and wait a second or two before speaking so they don't talk over each other and the quality was five by five, except for solar flares, sandstorms, rain. Things beyond my control. Thanks, Steve T -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TCP port, VPN and resolving the cutting voice problem
Any idea what is it about SIP over IAX2 that made such an improvement? -M On Thu, Dec 2, 2010 at 6:01 AM, Steve Totaro stot...@asteriskhelpdesk.com wrote: If getting a second circuit is out of the question. 1. Switch to SIP 2. Install and Learn Vyatta for QoS (Squid may help you quite a bit as well) as your router (or whatever you prefer) I use the paid versions of Vyatta but the free edition should be sufficient. I did the same setup over OpenVPN VSAT links in Iraq, 700ms ping times. I used GSM and some tricks on the Vyatta box. Originally, before I deployed the above, it was a wild west situation like what you have now. Going from G729 to GSM made a big improvement in conjunction with QoS. My theory on that is that G729 is already a very lossy codec, so any more loss, garbled audio. GSM is less lossy. Switch from IAX to SIP was another huge improvement, and then finally putting Vyatta and QoS as my router made calls almost crystal clear. There was the obvious lag time but users get used to that and wait a second or two before speaking so they don't talk over each other and the quality was five by five, except for solar flares, sandstorms, rain. Things beyond my control. Thanks, Steve T -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TCP port, VPN and resolving the cutting voice problem
No but if google my posts about IAX2, you will see that I have seen IAX2 cause so many problems with audio, I have made a good amount of money just switching customers to SIP. Even a large ITSP. I have found it to be responsible for poor audio in over a dozen cases and after switching to SIP, the audio was five by. Several people that work for Digium that will remain anonymous, have said to only use IAX when absolutely needed. You will also see people agreeing with me and others that have no issues. I just use SIP. Thanks, Steve T On Thu, Dec 2, 2010 at 9:27 AM, Mark Deneen mden...@gmail.com wrote: Any idea what is it about SIP over IAX2 that made such an improvement? -M On Thu, Dec 2, 2010 at 6:01 AM, Steve Totaro stot...@asteriskhelpdesk.com wrote: If getting a second circuit is out of the question. 1. Switch to SIP 2. Install and Learn Vyatta for QoS (Squid may help you quite a bit as well) as your router (or whatever you prefer) I use the paid versions of Vyatta but the free edition should be sufficient. I did the same setup over OpenVPN VSAT links in Iraq, 700ms ping times. I used GSM and some tricks on the Vyatta box. Originally, before I deployed the above, it was a wild west situation like what you have now. Going from G729 to GSM made a big improvement in conjunction with QoS. My theory on that is that G729 is already a very lossy codec, so any more loss, garbled audio. GSM is less lossy. Switch from IAX to SIP was another huge improvement, and then finally putting Vyatta and QoS as my router made calls almost crystal clear. There was the obvious lag time but users get used to that and wait a second or two before speaking so they don't talk over each other and the quality was five by five, except for solar flares, sandstorms, rain. Things beyond my control. Thanks, Steve T -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TCP port, VPN and resolving the cutting voice problem
Dear; I understood that Vyatta is the solution for the QoS, but I am not able to know if I can use a Vyatta hardware router to be DSL router and I set my QoS in it to resolve the voice problem. Is it possible? Thanks for the help. Regards Bilal Thanks all for ur participation and kindly advise. As I noticed that jitterbuffer could help if the ping does not have request time out but the voice is also cutting .. but in that case, I have to set the jitterbuffer at the IP Phones and Asterisk boxes. I have a polycom phone for example, and to set the jitterbuffer there are the following paramters: Payload Size Jitter Buffer Minimum Jitter Buffer Shrink Jitter Buffer Maximum When it use the minimum, and when it use the Shrink and when it use the maximum? If to look at the asterisk (in the SIP or IAX files) then there are a paramters for the jitterbuffer also, but really I am not able to know when to use this and when to use this: jenable, jbforce, jbmaxsize, jbresyncthreashold, jbimpl, jblog How to use the jbresyncthreashold? In which case? Regarding to the QoS, which will be need in case having a packet loose, correct? I just need to ask about something: What I will be able to do if my ISP did not setup the QoS at his side? What kind of settings I can do in my DSL router (in case of Cisco, or in case of Linksys that running linux firmware)? From the other side, if I used linux server to set the QoS, so do I have to let all the network elements to pass this linux server (so it will be the default gateway for other elements)? Appreciate the kindly help. Regards Bilal If getting a second circuit is out of the question. 1. Switch to SIP 2. Install and Learn Vyatta for QoS (Squid may help you quite a bit as well) as your router (or whatever you prefer) I use the paid versions of Vyatta but the free edition should be sufficient. I did the same setup over OpenVPN VSAT links in Iraq, 700ms ping times. I used GSM and some tricks on the Vyatta box. Originally, before I deployed the above, it was a wild west situation like what you have now. Going from G729 to GSM made a big improvement in conjunction with QoS. My theory on that is that G729 is already a very lossy codec, so any more loss, garbled audio. GSM is less lossy. Switch from IAX to SIP was another huge improvement, and then finally putting Vyatta and QoS as my router made calls almost crystal clear. There was the obvious lag time but users get used to that and wait a second or two before speaking so they don't talk over each other and the quality was five by five, except for solar flares, sandstorms, rain. Things beyond my control. Thanks, Steve T -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TCP port, VPN and resolving the cutting voice problem
Hi All; Can I run the IAX on TCP port instead of UDP port? If I ran IAX in TCP port, and in case my network was having a lot of users doing browse on the internet and downloading, so in that case and if the IAX used TCP port, so the voice will be better than using UDP (because in TCP the lost packets will be resend while in TCP it will not which will cause the voice to be cutting)? Same thing if we used the VPN, and in case of other users are using the Internet to do browsing and downloading then the voice quality will be better than without VPN as the VPN is using TCP? The internet bandwidth is not that small .. but the users are doing a big amount of work and we would like to overcome the packets losses in case of using the UDP as the packets are not resend. Any advise for this? What could be a solution that I can apply it to resolve the voice cutting if the Asterisk was using the internet that is shared with the users in the office that are doing download and browsing? One more thing, what about using the Buffering or any other technique that can help to overcome packet losses due to the internet download and browsing? Appreciate any help or advise? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TCP port, VPN and resolving the cutting voice problem
On 30 Nov 2010, at 09:28, bilal ghayyad wrote: If I ran IAX in TCP port, and in case my network was having a lot of users doing browse on the internet and downloading, so in that case and if the IAX used TCP port, so the voice will be better than using UDP (because in TCP the lost packets will be resend while in TCP it will not which will cause the voice to be cutting)? The re-sending would introduce massive latency and jitter. That's why UDP is used. In real-time voice, by the time the packet is 'missed' it's too late to retransmit it. S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TCP port, VPN and resolving the cutting voice problem
Dear; I know understand the latency due to the resending .. But if the link was have a good speed internet, then resending will make a big latency? Maybe this latency better than having a cutting voice? What if we reduce the packet size and make it TCP, so resending might cause acceptable delay? But again, what about running IAX in TCP port, this is possible? Any other solution to resolve the cutting in the voice while others doing download and browsing? Regards Bilal On 30 Nov 2010, at 09:28, bilal ghayyad wrote: If I ran IAX in TCP port, and in case my network was having a lot of users doing browse on the internet and downloading, so in that case and if the IAX used TCP port, so the voice will be better than using UDP (because in TCP the lost packets will be resend while in TCP it will not which will cause the voice to be cutting)? The re-sending would introduce massive latency and jitter. That's why UDP is used. In real-time voice, by the time the packet is 'missed' it's too late to retransmit it. S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TCP port, VPN and resolving the cutting voice problem
I know understand the latency due to the resending .. But if the link was have a good speed internet, then resending will make a big latency? I think the point is that with TCP, congestion will create a vicious circle of more congestion, while with UDP congestion is bad in itself, but doesn't result in more congestion created by the original congestion. That being said, isn't UDP sometimes looked at as being lower priority than TCP by many routers out there and dropped first when congestion does occur? That makes it a good reason to use TCP in some cases. Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TCP port, VPN and resolving the cutting voice problem
On Tue, Nov 30, 2010 at 4:28 AM, bilal ghayyad bilmar...@yahoo.com wrote: Hi All; Can I run the IAX on TCP port instead of UDP port? If I ran IAX in TCP port, and in case my network was having a lot of users doing browse on the internet and downloading, so in that case and if the IAX used TCP port, so the voice will be better than using UDP (because in TCP the lost packets will be resend while in TCP it will not which will cause the voice to be cutting)? Same thing if we used the VPN, and in case of other users are using the Internet to do browsing and downloading then the voice quality will be better than without VPN as the VPN is using TCP? The internet bandwidth is not that small .. but the users are doing a big amount of work and we would like to overcome the packets losses in case of using the UDP as the packets are not resend. Any advise for this? What could be a solution that I can apply it to resolve the voice cutting if the Asterisk was using the internet that is shared with the users in the office that are doing download and browsing? One more thing, what about using the Buffering or any other technique that can help to overcome packet losses due to the internet download and browsing? Appreciate any help or advise? Regards Bilal 1. Drop IAX and use SIP 2. Use some QoS or traffic management. There are plenty of opensource products, I go with Vyatta every time. 3. Get a dedicated VoIP pipe that will not be in contention with YouTube or whatever. Thanks, Steve T -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TCP port, VPN and resolving the cutting voice problem
On Tue, Nov 30, 2010 at 1:00 PM, Steve Totaro stot...@totarotechnologies.com wrote: On Tue, Nov 30, 2010 at 4:28 AM, bilal ghayyad bilmar...@yahoo.com wrote: Hi All; Can I run the IAX on TCP port instead of UDP port? If I ran IAX in TCP port, and in case my network was having a lot of users doing browse on the internet and downloading, so in that case and if the IAX used TCP port, so the voice will be better than using UDP (because in TCP the lost packets will be resend while in TCP it will not which will cause the voice to be cutting)? Same thing if we used the VPN, and in case of other users are using the Internet to do browsing and downloading then the voice quality will be better than without VPN as the VPN is using TCP? The internet bandwidth is not that small .. but the users are doing a big amount of work and we would like to overcome the packets losses in case of using the UDP as the packets are not resend. Any advise for this? What could be a solution that I can apply it to resolve the voice cutting if the Asterisk was using the internet that is shared with the users in the office that are doing download and browsing? One more thing, what about using the Buffering or any other technique that can help to overcome packet losses due to the internet download and browsing? Appreciate any help or advise? Regards Bilal 1. Drop IAX and use SIP 2. Use some QoS or traffic management. There are plenty of opensource products, I go with Vyatta every time. 3. Get a dedicated VoIP pipe that will not be in contention with YouTube or whatever. Thanks, Steve T I would suggest #3 because it is bullet proof unless your calls are maxing the bandwidth. You can always sell it to the decision makers as a business continuity contingency plan. The suits like those buzzwords and if the pipe is big enough, then you could allow mission critical business data to use that circuit. It is an easy sale to the bossmen, especially with the way you can talk down ISPs nowdays. Haggle with them, they are dying for the business. I got almost every circuit for half off the original quote. Just wait till the end of the month and say, I can have this singed and faxed over today if you can provide (name your terms, MRC, NRC, contract duration) It is a different world now. Americans are generally not very good hagglers. My travels have taught me many tricks and you can haggle virtually anything, within reason of course. Get quotes from all the carriers, haggle with them all, and then use them against each other, it can be time consuming, but I have paid for my salary in savings a few times over. Thanks, Steve T -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TCP port, VPN and resolving the cutting voice problem
On Tue, Nov 30, 2010 at 2:28 AM, bilal ghayyad bilmar...@yahoo.com wrote: If I ran IAX in TCP port, and in case my network was having a lot of users doing browse on the internet and downloading, so in that case and if the IAX used TCP port, so the voice will be better than using UDP (because in TCP the lost packets will be resend while in TCP it will not which will cause the voice to be cutting)? Not necessarily. See below. Basically the problem is that you have a congested link, and TCP is not the fix for congestion. Are you sure you are getting packet loss, and not just delayed packets, that might be arriving AFTER the jitter-buffer's max delay? Either would create the same symptom. But the solution to them is slightly different. Same thing if we used the VPN, and in case of other users are using the Internet to do browsing and downloading then the voice quality will be better than without VPN as the VPN is using TCP? TCP VPNs are bad for several reasons - namely that TCP inside TCP will generate excessive and unnecessary retransmissions. That's why most VPNs use UDP or IPSEC. TCP in TCP will increase delay and/or congestion on your links. The internet bandwidth is not that small .. but the users are doing a big amount of work and we would like to overcome the packets losses in case of using the UDP as the packets are not resend. Any advise for this? Yes. If you are using DSL/cable/other-commodity-circuit, I'd suggest a second DSL circuit to be used only for VoIP. Nobody likes to pay for that, I know, but that's really the solution. If you are using an (expensive) enterprise-class circuit (metro ethernet, DS3, OC3, etc) for internet, work with your provider. At the very least, have the provider does some form of fair queuing and you do the same, you'll probably eliminate 95% of your problem. If they are willing to do QoS to your specs, even better (but I wouldn't count on this). But clearly the way the circuit is configured today, you are having packet loss (the cutting out of voice) or excessive queing of packets. This is because queues in routers are getting too full, and something has to be dropped or something is arriving too late for the jitter buffers on the VoIP equipment to compensate. In otherwords, you are bandwidth constrained. So you need to either increase your bandwidth (expensive!) or implement QoS of some type. There are some ways to implement QoS on your end if your ISP won't cooperate, but it's not a 100% perfect solution. What could be a solution that I can apply it to resolve the voice cutting if the Asterisk was using the internet that is shared with the users in the office that are doing download and browsing? QoS. One more thing, what about using the Buffering or any other technique that can help to overcome packet losses due to the internet download and browsing? Certainly. If your problem is lost packets, you need QoS or bandwidth, but that aside, increased buffers in routers might help or hurt, depending on how things are behaving. You can try both (your ISP will need to do the same, if you are getting cut-outs on inbound packets; if you can get your ISP to adjust this, you can probably get him to just implement QoS and be done with this; If he can't implement QoS, at least get him to do some sort of fair queuing!). If your problem is excessively delayed (due to queuing) packets, you also need QoS or bandwidth. But you can increase the jitter buffer on both ends of the VoIP call. If you use a VoIP provider, they will need to increase the buffer size on their end. Of course this will increase the amount of talk-over and result in less user satisfaction. Delay is a bad thing on phone calls. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TCP port, VPN and resolving the cutting voice problem
I know understand the latency due to the resending .. But if the link was have a good speed internet, then resending will make a big latency? Maybe this latency better than having a cutting voice? Fundamentally, TCP's congestion-avoidance and loss-recovery logic simply won't work well with voice, unless you're willing to accept a really horrendous latency (hundreds of milliseconds) and then perhaps not even then. TCP is designed to ensure reliable data delivery and reasonable network efficiency (i.e. avoiding congestion and avoiding an excessive number of retransmissions) and is simply not well suited for isochronous (or close-to-isochronous) data streams such as VoIP. What if we reduce the packet size and make it TCP, so resending might cause acceptable delay? But again, what about running IAX in TCP port, this is possible? Sure, it is *possible*. I don't think anyone has implemented it, because everyone who might is probably pretty well aware that it would not work well. Any other solution to resolve the cutting in the voice while others doing download and browsing? I'd recommend the following general approach (not my own ideas, just ones I've adopted from other peoples' recommendations): - Deliberately throttle both inbound and outbound TCP connections, so that they do not consume all of your link bandwidth. Set aside some amount of the link bandwidth for VoIP traffic (SIP or IAX2) traveling over UDP. For outbound traffic, what you need is a rate-limiting traffic shaper which supports multiple queues. Linux can do this with its advanced traffic shaping modules. For inbound traffic, what you need to do is prevent the sending TCP stack (at the far end) from being able to queue up and transmit an excessively large amount of traffic. Since you have no *direct* control over the remote systems, you have to do it indirectly... and the way you do it is by input policing. This simply means that when incoming TCP packets start consuming more than a specific percentage of your inbound bandwidth, you start dropping them... artificially creating a lost packet error, which then causes the sending system to reduce its transmit window and enter a congestion-avoidance process. This also can be done using the Linux traffic-shaping modules. - Prioritize the packets you send, with VoIP packets being transmitted before TCP packets. This can be done using a combination of traffic-shaping modules (to set up and prioritize the queues and set their transmit rates) and iptables (which can be used to mark VoIP packets as needing expedited delivery). Take a look at http://lartc.org/howto/ - it's a complex subject but a well-designed traffic shaping configuration can have really excellent benefits. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
