Re: [Asterisk-Users] QoS for improvements
I've spent may hours to play with HTB QoS settings on the firewall, but with absolutely no effect. In fact, this is normal, because the time required to let a data packet going through the ADSL line will break the voice jitter. The only right way to handle this issue is to modify the MTU on the router. Without setting a TOS for voip, data where going through and voice was unusable. With a lowdelay (0x10) TOS set for voip, voice was going through, but data was blocked. With a lowdelay TOS and an HTB QoS on the router, data where going through slowly and voice was scambled. After many tests, an MTU of 700 did work quite well. I did loose 15% of bandwidth for data (twice more overheads), but data and voice may be used together. Those tests have been done on a 256 kbps up stream. There is a quite good explenation about this issue on Cisco's web site, and about they're LFI technology (link fragmentation and interleaving): http://www.cisco.com/warp/public/788/voice-qos/voip-mlppp.html#link_frag Jean-Chrsitophe Kumara Jayaweera a écrit : Hello! Everybody!!, I want to run VoIP in the same LAN (15 windows clients) which we use for surfing the Internet. 6-7 softphones in the same client's machines is 'the target'. My DSL is 128kbps, (I can go to 256kbps if required). So, I am told to install some QoS's in the LAN to improve the voice quality. Frankly, I don't know what it (QoS= Quality of Service) is. I hope you may help me giving Links to read and briefing me your ideas. Thanks to everybody in the list. So far my success and progress are your help. Thanks again Kumara ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] QoS for improvements
Sometimes this all sounds so complicatedbut it needn't be. I suppose it can vary with the size of your installation. I use a m0n0wall router on a Covad DSL line. Using m0n0's traffic shaping feature I establish inbound and outbound pipes which are bandwidth restricted to just less than my mesured average DSL rate. I then break my traffic into three priority ques in each direction; highest priority, medium priority, low priority. I assign all IAX traffic in/out to the highest priority que, and map all IAX ports to the * server inside the LAN. In fact, I just used the Magic Shaper Wizard in m0n0, then added IAX specific entries to give it highest priority. The whole process took about a half hour. Just as easy as the Linksys BEFSR-81 that I had before, but more reliable and more controllable. Now to be fair, I only have about 6 phones. I use IAX2 to all my ITSPs and SIP in-house only. My DSL is 3M down / 768k up. Michael On Fri, 06 May 2005 08:37:20 +0200, Jean-Christophe Heger wrote: I've spent may hours to play with HTB QoS settings on the firewall, but with absolutely no effect. In fact, this is normal, because the time required to let a data packet going through the ADSL line will break the voice jitter. The only right way to handle this issue is to modify the MTU on the router. Without setting a TOS for voip, data where going through and voice was unusable. With a lowdelay (0x10) TOS set for voip, voice was going through, but data was blocked. With a lowdelay TOS and an HTB QoS on the router, data where going through slowly and voice was scambled. After many tests, an MTU of 700 did work quite well. I did loose 15% of bandwidth for data (twice more overheads), but data and voice may be used together. Those tests have been done on a 256 kbps up stream. There is a quite good explenation about this issue on Cisco's web site, and about they're LFI technology (link fragmentation and interleaving): http://www.cisco.com/warp/public/788/voice-qos/voip-mlppp.html#link_frag Jean-Chrsitophe Kumara Jayaweera a écrit : Hello! Everybody!!, I want to run VoIP in the same LAN (15 windows clients) which we use for surfing the Internet. 6-7 softphones in the same client's machines is 'the target'. My DSL is 128kbps, (I can go to 256kbps if required). So, I am told to install some QoS's in the LAN to improve the voice quality. Frankly, I don't know what it (QoS= Quality of Service) is. I hope you may help me giving Links to read and briefing me your ideas. Thanks to everybody in the list. So far my success and progress are your help. Thanks again Kumara ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] QoS for improvements
That's funny, people having good bandwidth always have a better way to do it. You should feel lucky, because no one provides 768kbps upstreams in Switzerland, except if you want to pay 1'000 USD per month for a leased line. There is nothing complicated, just mathematics. Here is the formula: MTU: Maximum transmit unit = 1492 Bytes (ADSL) UP: Up stream t: time spent for a full framed packet (1492 Bytes) t = 8 * UP / MTU 128k upstream - 91 ms 256k upstream - 45 ms 512k upstream - 23 ms 768k upstream - 15 ms Using a codec, such as GSM or G.729, will take around 20 to 30 ms for encoding and decoding. While you wait for a full framed packet to go through the ADSL line, a voip packet will wait need 20 ms for encoding, 91 ms for waiting (at the most), 30 ms to go to the destination (at the best), and 10 ms to be decoded. Total: 20+91+30+20 = 161 ms. Even worse, the jitter balance will play around 100 ms, because of waiting on full framed packed. That's what I call breaking the jitter, because not all equipment does support such jitters. Depending on the line and the distance (hops), you can easily add 50 ms, bringing the total around 200 ms. Therefore we consider a conversation as good, when the overall delay is under 150 ms. Bringing the MTU to 700 ms does bring back the overall delay to this target, and the jitter to 50 ms. Regarding the results, 768 kbps up stream is working even without QoS ( 100 ms). PFIFO (TOS=lowdelay) is good enough for perfect communications. So, any other magical solution ? Jean-Christophe Michael Graves a écrit : Sometimes this all sounds so complicatedbut it needn't be. I suppose it can vary with the size of your installation. I use a m0n0wall router on a Covad DSL line. Using m0n0's traffic shaping feature I establish inbound and outbound pipes which are bandwidth restricted to just less than my mesured average DSL rate. I then break my traffic into three priority ques in each direction; highest priority, medium priority, low priority. I assign all IAX traffic in/out to the highest priority que, and map all IAX ports to the * server inside the LAN. In fact, I just used the Magic Shaper Wizard in m0n0, then added IAX specific entries to give it highest priority. The whole process took about a half hour. Just as easy as the Linksys BEFSR-81 that I had before, but more reliable and more controllable. Now to be fair, I only have about 6 phones. I use IAX2 to all my ITSPs and SIP in-house only. My DSL is 3M down / 768k up. Michael On Fri, 06 May 2005 08:37:20 +0200, Jean-Christophe Heger wrote: ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] QoS for improvements
Jean-Christophe, Thank you for the explanation. I've never been in a situation demanding adjustment of MTU. It's not so much that I think I have a better way, only that my circumstances lend themselves to a simple solution. I did start out using * with only 256k upload speed. I decided to stay with G.711 and purchase the better connection, since it was available. In your area where raw bandwidth is costly is there any sense in using ISDN lines instead of ADSL? I'd love to dump my SBC POTS lines and get two BRIs, but BRI capable hardware meeting US standards is scarce/non-existent. Michael On Fri, 06 May 2005 18:34:00 +0200, Jean-Christophe Heger wrote: That's funny, people having good bandwidth always have a better way to do it. You should feel lucky, because no one provides 768kbps upstreams in Switzerland, except if you want to pay 1'000 USD per month for a leased line. There is nothing complicated, just mathematics. Here is the formula: MTU: Maximum transmit unit = 1492 Bytes (ADSL) UP: Up stream t: time spent for a full framed packet (1492 Bytes) t = 8 * UP / MTU 128k upstream - 91 ms 256k upstream - 45 ms 512k upstream - 23 ms 768k upstream - 15 ms Using a codec, such as GSM or G.729, will take around 20 to 30 ms for encoding and decoding. While you wait for a full framed packet to go through the ADSL line, a voip packet will wait need 20 ms for encoding, 91 ms for waiting (at the most), 30 ms to go to the destination (at the best), and 10 ms to be decoded. Total: 20+91+30+20 = 161 ms. Even worse, the jitter balance will play around 100 ms, because of waiting on full framed packed. That's what I call breaking the jitter, because not all equipment does support such jitters. Depending on the line and the distance (hops), you can easily add 50 ms, bringing the total around 200 ms. Therefore we consider a conversation as good, when the overall delay is under 150 ms. Bringing the MTU to 700 ms does bring back the overall delay to this target, and the jitter to 50 ms. Regarding the results, 768 kbps up stream is working even without QoS ( 100 ms). PFIFO (TOS=lowdelay) is good enough for perfect communications. So, any other magical solution ? Jean-Christophe Michael Graves a écrit : Sometimes this all sounds so complicatedbut it needn't be. I suppose it can vary with the size of your installation. I use a m0n0wall router on a Covad DSL line. Using m0n0's traffic shaping feature I establish inbound and outbound pipes which are bandwidth restricted to just less than my mesured average DSL rate. I then break my traffic into three priority ques in each direction; highest priority, medium priority, low priority. I assign all IAX traffic in/out to the highest priority que, and map all IAX ports to the * server inside the LAN. In fact, I just used the Magic Shaper Wizard in m0n0, then added IAX specific entries to give it highest priority. The whole process took about a half hour. Just as easy as the Linksys BEFSR-81 that I had before, but more reliable and more controllable. Now to be fair, I only have about 6 phones. I use IAX2 to all my ITSPs and SIP in-house only. My DSL is 3M down / 768k up. Michael On Fri, 06 May 2005 08:37:20 +0200, Jean-Christophe Heger wrote: ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] QoS for improvements
Yeah, agree with that, but almost the provided upstream is not guaranteed (except you have lease lines, and Pay 1'000's UDS per month). Yes, the g729 codex is a good solution but not for a large number of users /callers. A. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jean-Christophe Heger Sent: Friday, 06 May 2005 18:34 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] QoS for improvements That's funny, people having good bandwidth always have a better way to do it. You should feel lucky, because no one provides 768kbps upstreams in Switzerland, except if you want to pay 1'000 USD per month for a leased line. There is nothing complicated, just mathematics. Here is the formula: MTU: Maximum transmit unit = 1492 Bytes (ADSL) UP: Up stream t: time spent for a full framed packet (1492 Bytes) t = 8 * UP / MTU 128k upstream - 91 ms 256k upstream - 45 ms 512k upstream - 23 ms 768k upstream - 15 ms Using a codec, such as GSM or G.729, will take around 20 to 30 ms for encoding and decoding. While you wait for a full framed packet to go through the ADSL line, a voip packet will wait need 20 ms for encoding, 91 ms for waiting (at the most), 30 ms to go to the destination (at the best), and 10 ms to be decoded. Total: 20+91+30+20 = 161 ms. Even worse, the jitter balance will play around 100 ms, because of waiting on full framed packed. That's what I call breaking the jitter, because not all equipment does support such jitters. Depending on the line and the distance (hops), you can easily add 50 ms, bringing the total around 200 ms. Therefore we consider a conversation as good, when the overall delay is under 150 ms. Bringing the MTU to 700 ms does bring back the overall delay to this target, and the jitter to 50 ms. Regarding the results, 768 kbps up stream is working even without QoS ( 100 ms). PFIFO (TOS=lowdelay) is good enough for perfect communications. So, any other magical solution ? Jean-Christophe Michael Graves a écrit : Sometimes this all sounds so complicatedbut it needn't be. I suppose it can vary with the size of your installation. I use a m0n0wall router on a Covad DSL line. Using m0n0's traffic shaping feature I establish inbound and outbound pipes which are bandwidth restricted to just less than my mesured average DSL rate. I then break my traffic into three priority ques in each direction; highest priority, medium priority, low priority. I assign all IAX traffic in/out to the highest priority que, and map all IAX ports to the * server inside the LAN. In fact, I just used the Magic Shaper Wizard in m0n0, then added IAX specific entries to give it highest priority. The whole process took about a half hour. Just as easy as the Linksys BEFSR-81 that I had before, but more reliable and more controllable. Now to be fair, I only have about 6 phones. I use IAX2 to all my ITSPs and SIP in-house only. My DSL is 3M down / 768k up. Michael On Fri, 06 May 2005 08:37:20 +0200, Jean-Christophe Heger wrote: ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] QoS for improvements
don't know what it (QoS= Quality of Service) is. I hope you may help me giving Links to read and briefing me your ideas. 1 minute of google search I found this : http://compnetworking.about.com/od/networkdesign/l/bldef_qos.htm which looks like a pretty nice explanation hth ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] QoS for improvements
Kumara Jayaweera wrote: I want to run VoIP in the same LAN (15 windows clients) which we use for surfing the Internet. Some magic words: QoS Asterisk HTB TC. Not easy to find good material over the internet, but Google may give you some ideas - how to use them is another problem, which you have to figure out alone, as there are a few resources to research. Start here: http://www.krisk.org/astlinux/misc/astshape ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
