[asterisk-users] QoS : tos and cos settings
Hi All, I need some assistance with QoS. We have multiple Asterisk servers on a MPLS network. We have just moved over to Verizon and for us to get QoS Verizon are saying we need to use af41. I need to check what exactly I need to do as this is a new area for me. We only use IAX over our WAN links as SIP is on the local LAN. In IAX I have set: /etc/asterisk/iax.conf tos=af41 cos=4 https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service this pages sais I need to use a vconfig command. This is what is confusing me. vconfig set_egress_map [vlan-device] [skb-priority] [vlan-qos] is someone able to suggest what I should use here? I'm guessing my iax.conf setting is correct. Your help is greatly appreciated. Regards David-- _ -- 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] QoS : tos and cos settings
On 12-07-15 10:23 PM, Klaverstyn, David C wrote: Hi All, I need some assistance with QoS. We have multiple Asterisk servers on a MPLS network. We have just moved over to Verizon and for us to get QoS Verizon are saying we need to use af41. Rather than get the application to do this, get Linux to do it right at the network layer (assuming you're running Linux). iptables -A OUTPUT -t mangle -p udp -dport 4569 -j DSCP --set-dscp-class AF41 Then do the same thing on the other end, and you should be golden! -- Looking for (employment|contract) work in the Internet industry, preferrably working remotely. Building / Supporting the net since 2400 baud was the hot thing. Ask for a resume! ispbuil...@gmail.com -- _ -- 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] QoS and Asterisk
I have discussed QoS with our ISP and in order to implement this, I need to make sure all VoIP packets are marked in the IP packet header (IPP bits?). Does Asterisk automatically marks the VoIP packets or do I need to do something in Asterisk? I need to do this for SIP and H323 protocols. Any information would be helpful. Thanks, Hin -- _ -- 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] QoS and Asterisk
On 07/15/2010 11:13 AM, hin lee wrote: I have discussed QoS with our ISP and in order to implement this, I need to make sure all VoIP packets are marked in the IP packet header (IPP bits?). Does Asterisk automatically marks the VoIP packets or do I need to do something in Asterisk? I need to do this for SIP and H323 protocols. Any information would be helpful. Thanks, Hin Have you looked in /etc/asterisk/sip.conf ? -- _ -- 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] QOS/DSCP for IAX?
Is it possible to set QOS/COS/DSCP on IAX packets? I see some parameters in sip.conf but not iax.conf Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QOS/DSCP for IAX?
I actually see the TOS setting in iax.conf, but the default (commented out) is EF - which doesn't even match a valid bit combination according to voip-info wiki If this is the right place, what TOS value are people using succesfully over an ADSL connection? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis Sent: Thursday, October 01, 2009 2:27 PM To: Asterisk Users List Subject: [asterisk-users] QOS/DSCP for IAX? Is it possible to set QOS/COS/DSCP on IAX packets? I see some parameters in sip.conf but not iax.conf Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QOS/DSCP for IAX?
Did you look at this wiki - http://www.voip-info.org/wiki/index.php?page=Asterisk+config+iax.conf ? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis Sent: Thursday, October 01, 2009 1:36 PM To: 'Asterisk Users List' Subject: Re: [asterisk-users] QOS/DSCP for IAX? I actually see the TOS setting in iax.conf, but the default (commented out) is EF - which doesn't even match a valid bit combination according to voip-info wiki If this is the right place, what TOS value are people using succesfully over an ADSL connection? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis Sent: Thursday, October 01, 2009 2:27 PM To: Asterisk Users List Subject: [asterisk-users] QOS/DSCP for IAX? Is it possible to set QOS/COS/DSCP on IAX packets? I see some parameters in sip.conf but not iax.conf Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QOS/DSCP for IAX?
Michelle Dupuis wrote: I actually see the TOS setting in iax.conf, but the default (commented out) is EF - which doesn't even match a valid bit combination according to voip-info wiki If this is the right place, what TOS value are people using succesfully over an ADSL connection? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis Sent: Thursday, October 01, 2009 2:27 PM To: Asterisk Users List Subject: [asterisk-users] QOS/DSCP for IAX? Is it possible to set QOS/COS/DSCP on IAX packets? I see some parameters in sip.conf but not iax.conf Thanks Yes the tos setting is the right place and EF is an acceptable value. EF is the differentiated services code point (or dscp) for expedited forwarding. The sample sip.conf defaults tos_audio to EF as well. The iax.conf wiki page only shows the old type of service values which are considered deprecated. Look at this page for more info on diffserv: http://www.voip-info.org/wiki/view/DiffServ As for what to use, well, that depends on whether your upstream provider even honors what you set. They may use the old type of service values, they may use dscp or they may ignore what you put there entirely. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QOS/DSCP for IAX?
That link is great thanks. From what I read elsewhere, ToS is just the first 3 bits which should be honored by DSCP (first 5 bits)- even old equip should be DSCP compatible...or I need to do more reading :) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dave Fullerton Sent: Thursday, October 01, 2009 3:01 PM To: Asterisk Users List Subject: Re: [asterisk-users] QOS/DSCP for IAX? Michelle Dupuis wrote: I actually see the TOS setting in iax.conf, but the default (commented out) is EF - which doesn't even match a valid bit combination according to voip-info wiki If this is the right place, what TOS value are people using succesfully over an ADSL connection? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis Sent: Thursday, October 01, 2009 2:27 PM To: Asterisk Users List Subject: [asterisk-users] QOS/DSCP for IAX? Is it possible to set QOS/COS/DSCP on IAX packets? I see some parameters in sip.conf but not iax.conf Thanks Yes the tos setting is the right place and EF is an acceptable value. EF is the differentiated services code point (or dscp) for expedited forwarding. The sample sip.conf defaults tos_audio to EF as well. The iax.conf wiki page only shows the old type of service values which are considered deprecated. Look at this page for more info on diffserv: http://www.voip-info.org/wiki/view/DiffServ As for what to use, well, that depends on whether your upstream provider even honors what you set. They may use the old type of service values, they may use dscp or they may ignore what you put there entirely. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QOS/DSCP for IAX?
Yea, kinda, sorta. The DSCP is six bits, which occupy six of the 8 bits in what is/was the type of service byte in an IP packet. Three of the 6 DSCP bits reside over the old precedence field and three reside over the old low delay, high throughput and high reliability fields (those three often referred to as TOS). The DSCP code points are designed to be backwards compatible with the PRECEDENCE portion of the old tos. The low delay, high throughput and high reliability bits have been redefined and no longer are backwards compatible. When doing my research I found some web sites displayed the tos byte in different bit-orders (cisco with precedence first, wikipedia with precedence last). It was confusing as heck. I also have some old equipment that does not understand DSCP/Diffserv. What I ended up doing was making asterisk and phones use the dscp code points and my old router software queue packets based on what it sees in the precedence field. Works like a charm. Good luck. -Dave Michelle Dupuis wrote: That link is great thanks. From what I read elsewhere, ToS is just the first 3 bits which should be honored by DSCP (first 5 bits)- even old equip should be DSCP compatible...or I need to do more reading :) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dave Fullerton Sent: Thursday, October 01, 2009 3:01 PM To: Asterisk Users List Subject: Re: [asterisk-users] QOS/DSCP for IAX? Michelle Dupuis wrote: I actually see the TOS setting in iax.conf, but the default (commented out) is EF - which doesn't even match a valid bit combination according to voip-info wiki If this is the right place, what TOS value are people using succesfully over an ADSL connection? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis Sent: Thursday, October 01, 2009 2:27 PM To: Asterisk Users List Subject: [asterisk-users] QOS/DSCP for IAX? Is it possible to set QOS/COS/DSCP on IAX packets? I see some parameters in sip.conf but not iax.conf Thanks Yes the tos setting is the right place and EF is an acceptable value. EF is the differentiated services code point (or dscp) for expedited forwarding. The sample sip.conf defaults tos_audio to EF as well. The iax.conf wiki page only shows the old type of service values which are considered deprecated. Look at this page for more info on diffserv: http://www.voip-info.org/wiki/view/DiffServ As for what to use, well, that depends on whether your upstream provider even honors what you set. They may use the old type of service values, they may use dscp or they may ignore what you put there entirely. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QoS
At 06:37 AM 7/15/2009, you wrote: Ours is just internal, but the concept should be the same. My boss could talk on his phone fine until he cranked up Foxnews feed. Once the video started, he couldn't talk on his phone anymore (bad quality or total loss of call). What I've done here is probably a bit extreme, but we've never had a problem of any kind with our VOIP calls. it's a house, no more than 2 calls at a time on cable internet so it might just be that the connection is significantly faster than we ever use. I have a Linksys router that has the Asterisk box connected to a port marked High and the rest of the house is a second router connected to a port on the first router flagged as low or regular. I ran separate Cat5 for the phones and the computers. If I knew what I was doing I'd get a Linux box with 3 ports and have it do everything. Ira ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QoS
In my shop, we got a better router to support QOS and configured our Polycom phones to always request highest levels (UDP gets 6, everything else gets 3). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Tuesday, July 14, 2009 5:04 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] QoS Howdy, Getting ready to play with QoS settings. We have an asterisk 1.4.23 server running in a colo bunker in the US Virgin Islands under a large radio tower. That tower has multiple sector radio/antenna pairs that blanket a valley in 802.11a. The customers have directed dishes aimed at the sector antennas, mounted on their roofs. This setup has been working great for their broadband access for many years. Now we want to sell voice services on top of this infrastructure, and it works fine too, until they start some data intensive process on the customer end, like bittorrent :) We would like to avoid these problems by properly setting up packet prioritization between the customer and the sector radios, which we have control over. Any links to share to get us started? Basically from zero? :) Cheers, j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QoS
On Wed, 15 Jul 2009, Danny Nicholas wrote: In my shop, we got a better router to support QOS and configured our Polycom phones to always request highest levels (UDP gets 6, everything else gets 3). Did this apply to your connection to the net, or just internally? I am most concerned with the link between the customer premise and the next hop router, which is the slowest link in the path. We pay dearly for bandwidth down here, so most customers have only a 256Kbps radio link. Should be plenty for VoIP, and it is, until they start using it for something else at the same time. Cheers, j -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Tuesday, July 14, 2009 5:04 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] QoS Howdy, Getting ready to play with QoS settings. We have an asterisk 1.4.23 server running in a colo bunker in the US Virgin Islands under a large radio tower. That tower has multiple sector radio/antenna pairs that blanket a valley in 802.11a. The customers have directed dishes aimed at the sector antennas, mounted on their roofs. This setup has been working great for their broadband access for many years. Now we want to sell voice services on top of this infrastructure, and it works fine too, until they start some data intensive process on the customer end, like bittorrent :) We would like to avoid these problems by properly setting up packet prioritization between the customer and the sector radios, which we have control over. Any links to share to get us started? Basically from zero? :) Cheers, j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QoS
Ours is just internal, but the concept should be the same. My boss could talk on his phone fine until he cranked up Foxnews feed. Once the video started, he couldn't talk on his phone anymore (bad quality or total loss of call). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Wednesday, July 15, 2009 8:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] QoS On Wed, 15 Jul 2009, Danny Nicholas wrote: In my shop, we got a better router to support QOS and configured our Polycom phones to always request highest levels (UDP gets 6, everything else gets 3). Did this apply to your connection to the net, or just internally? I am most concerned with the link between the customer premise and the next hop router, which is the slowest link in the path. We pay dearly for bandwidth down here, so most customers have only a 256Kbps radio link. Should be plenty for VoIP, and it is, until they start using it for something else at the same time. Cheers, j -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Tuesday, July 14, 2009 5:04 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] QoS Howdy, Getting ready to play with QoS settings. We have an asterisk 1.4.23 server running in a colo bunker in the US Virgin Islands under a large radio tower. That tower has multiple sector radio/antenna pairs that blanket a valley in 802.11a. The customers have directed dishes aimed at the sector antennas, mounted on their roofs. This setup has been working great for their broadband access for many years. Now we want to sell voice services on top of this infrastructure, and it works fine too, until they start some data intensive process on the customer end, like bittorrent :) We would like to avoid these problems by properly setting up packet prioritization between the customer and the sector radios, which we have control over. Any links to share to get us started? Basically from zero? :) Cheers, j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QoS
Danny Nicholas wrote: Ours is just internal, but the concept should be the same. My boss could talk on his phone fine until he cranked up Foxnews feed. Therein lies the problem One should NOT contaminate their network with Fixed News ( AKA as Fox Noise ) In addition to overloading internal networks, it promotes brain rot. Peg Leg O'Brien Once the video started, he couldn't talk on his phone anymore (bad quality or total loss of call). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Wednesday, July 15, 2009 8:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] QoS On Wed, 15 Jul 2009, Danny Nicholas wrote: In my shop, we got a better router to support QOS and configured our Polycom phones to always request highest levels (UDP gets 6, everything else gets 3). Did this apply to your connection to the net, or just internally? I am most concerned with the link between the customer premise and the next hop router, which is the slowest link in the path. We pay dearly for bandwidth down here, so most customers have only a 256Kbps radio link. Should be plenty for VoIP, and it is, until they start using it for something else at the same time. Cheers, j -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Tuesday, July 14, 2009 5:04 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] QoS Howdy, Getting ready to play with QoS settings. We have an asterisk 1.4.23 server running in a colo bunker in the US Virgin Islands under a large radio tower. That tower has multiple sector radio/antenna pairs that blanket a valley in 802.11a. The customers have directed dishes aimed at the sector antennas, mounted on their roofs. This setup has been working great for their broadband access for many years. Now we want to sell voice services on top of this infrastructure, and it works fine too, until they start some data intensive process on the customer end, like bittorrent :) We would like to avoid these problems by properly setting up packet prioritization between the customer and the sector radios, which we have control over. Any links to share to get us started? Basically from zero? :) Cheers, j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Dog is my co-pilot ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QoS
On Wed, 2009-07-15 at 08:10 -0500, Danny Nicholas wrote: In my shop, we got a better router to support QOS and configured our Polycom phones to always request highest levels (UDP gets 6, everything else gets 3). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Tuesday, July 14, 2009 5:04 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] QoS Howdy, Getting ready to play with QoS settings. We have an asterisk 1.4.23 server running in a colo bunker in the US Virgin Islands under a large radio tower. That tower has multiple sector radio/antenna pairs that blanket a valley in 802.11a. The customers have directed dishes aimed at the sector antennas, mounted on their roofs. This setup has been working great for their broadband access for many years. Now we want to sell voice services on top of this infrastructure, and it works fine too, until they start some data intensive process on the customer end, like bittorrent :) We would like to avoid these problems by properly setting up packet prioritization between the customer and the sector radios, which we have control over. Any links to share to get us started? Basically from zero? :) snip I'm entirely unfamiliar with your environment and very new to Asterisk so please take what I say with a large dose of skepticism. We elected to move CoS right to the core switch and tried to keep it consistent throughout the path. Our environment is still pretty simple so we are using HP Procurve 2810 switches. Asterisk sits on its own VLAN. We believe we found some conflict between typical DSCP settings and Linux routers / firewalls in their default state. We initially set our systems to use Expedited Forwarding for both SIP and audio RTP. I believe this is b8 in Asterisk and 184 for our Snom phones. This sets the bits in the DSCP field as 101110. However, the default Linux packet prioritization (pfifo_fast) is looking at only the last three bits of that field (because it is not actually using DSCP but the ToS bits). It sees 110 and that middle 1 causes it to place the packets in band1 which is the default processing rather than band0 which is priority processing. We thus changed the DSCP header to 101100 (b0 in Asterisk and 176 in Snom). We believe this will cause default Linux routers / firewalls using pfifo_fast to process these packets in band0 (high priority). We then returned to our switch and told it to map DSCP header 101100 to its highest priority path. Thus, we should now have consistent CoS from the servers to the switches to the firewalls to the phones. Since our connection to Asterisk is via VPN, we also ensure the ToS bits are passed to the VPN header (be it IPSec or OpenVPN). I know that sounds dreadfully complicated but that is how we did it. If someone sees a better way or if we are unnecessarily complicating it, please let us know. If you need more information, I can probably post some of our internal documentation. Hope this helps - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] QoS
Howdy, Getting ready to play with QoS settings. We have an asterisk 1.4.23 server running in a colo bunker in the US Virgin Islands under a large radio tower. That tower has multiple sector radio/antenna pairs that blanket a valley in 802.11a. The customers have directed dishes aimed at the sector antennas, mounted on their roofs. This setup has been working great for their broadband access for many years. Now we want to sell voice services on top of this infrastructure, and it works fine too, until they start some data intensive process on the customer end, like bittorrent :) We would like to avoid these problems by properly setting up packet prioritization between the customer and the sector radios, which we have control over. Any links to share to get us started? Basically from zero? :) Cheers, j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QoS VPN
Despite the VPN overhead, running VOIP through VPN is good idea because VPN reorders encapsulated UDP packets in correct order. Security matters as well. I'd suggest to route VNC packets rather over internet than VPN (so do I), as VPN usually has the highest priority. On Thu, May 7, 2009 at 11:33 PM, Roberto Piola roberto.pi...@visiant.itwrote: I do not have examples, but if you are using the 1700 series router in order to originate the ipsec vpn, you may use command qos pre-classify (please search for it on cco.cisco.com) On Thu, May 7, 2009 at 9:54 PM, Brent Davidson br...@texascountrytitle.com wrote: I've got multiple satellite office all linked back to the main office via VPN. Each office has their own asterisk server which registers back to the main office's Asterisk server. Each office also has a 1Mb downstream / 384k - 768k upstream connection. The branches are using Speex for their connections back to the main office. The issue I'm having is that there are times that I need to VNC in to machines at the various offices for tech support while the user is also on the phone. Unfortunately the VNC connection apparently takes priority and makes it impossible for me to understand anything the person on the phone is saying, although they can still hear me fine. Our Main office uses a Cisco PIX 506 for the main firewall and VPN concentrator. Each branch office used a Cisco 1700 series router with IPSec enabled in the IOS. Is there any sort of QoS I can turn on on the main router or the branch routers to make sure the voice quality takes precedence over the VNC? (Any example configs would be greatly appreciated) Would I be better off routing the voice packets over the internet rather than the VPN, and could I safely do that without exposing the asterisk boxes to unnecessary security risks? (At present all of our asterisk boxes are behind the firewalls and only talk to each other over the VPN. All PSTN connection is done through TDM boards so they have no direct exposure to the internet.) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mvh, Aurimas Skirgaila ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QoS VPN
On Fri, 8 May 2009, Aurimas Skirgaila wrote: Despite the VPN overhead, running VOIP through VPN is good idea because VPN reorders encapsulated UDP packets in correct order. Security matters as well. Reorders? How so? I think it will maintain the order, only if they have arrived in the correct order. I'd suggest to route VNC packets rather over internet than VPN (so do I), as VPN usually has the highest priority. Unless QoS is implemented packets are first come first served. There is no usually has the highest priority. Routing one over the Internet versus over the VPN won't change that priority. j On Thu, May 7, 2009 at 11:33 PM, Roberto Piola roberto.pi...@visiant.itwrote: I do not have examples, but if you are using the 1700 series router in order to originate the ipsec vpn, you may use command qos pre-classify (please search for it on cco.cisco.com) On Thu, May 7, 2009 at 9:54 PM, Brent Davidson br...@texascountrytitle.com wrote: I've got multiple satellite office all linked back to the main office via VPN. Each office has their own asterisk server which registers back to the main office's Asterisk server. Each office also has a 1Mb downstream / 384k - 768k upstream connection. The branches are using Speex for their connections back to the main office. The issue I'm having is that there are times that I need to VNC in to machines at the various offices for tech support while the user is also on the phone. Unfortunately the VNC connection apparently takes priority and makes it impossible for me to understand anything the person on the phone is saying, although they can still hear me fine. Our Main office uses a Cisco PIX 506 for the main firewall and VPN concentrator. Each branch office used a Cisco 1700 series router with IPSec enabled in the IOS. Is there any sort of QoS I can turn on on the main router or the branch routers to make sure the voice quality takes precedence over the VNC? (Any example configs would be greatly appreciated) Would I be better off routing the voice packets over the internet rather than the VPN, and could I safely do that without exposing the asterisk boxes to unnecessary security risks? (At present all of our asterisk boxes are behind the firewalls and only talk to each other over the VPN. All PSTN connection is done through TDM boards so they have no direct exposure to the internet.) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mvh, Aurimas Skirgaila ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QoS VPN
Access-list 100 permit ip host asterisk server any Class-map match-any voip Match access-group 100 Policy-map voip Class voip Priority 256 Class class-default Fair-queue Interface fastethernet 0 Service-policy output voip Above is what I do to prioritize 256kbit of outbound bandwidth to voip calls, adjust accordingly. You must also use the qos pre-classify in your ipsec tunnel definitions for this to work, but it does work well. I know I'm potentially mapping other traffic than voip, but I'm lazy and don't want to classify the rtp and sip and iax ports, rarely does the box do any other traffic than voip as updates occur in off hours. You'll probably additionally want to match your ipsec keying traffic and give it priority bandwidth, if you're going to push voip through the tunnel you'll find yourself rekeying more often and want to make sure on a saturated link it gets priority so the tunnels don't drop. If you're on DSL, you probably want to research cascading the Qos, have a root policy that throttles all bandwidth to a certain speed, then a child policy that prioritizes that bandwidth, so you don't saturate your outbound circuit(think in terms of P2P protections). This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QoS VPN
On Fri, May 8, 2009 at 3:45 PM, Jeff LaCoursiere j...@jeff.net wrote: On Fri, 8 May 2009, Aurimas Skirgaila wrote: Despite the VPN overhead, running VOIP through VPN is good idea because VPN reorders encapsulated UDP packets in correct order. Security matters as well. Reorders? How so? I think it will maintain the order, only if they have arrived in the correct order. UDP doesn't guarantee that over long way packets arrive in correct order, while TCP based VPN would sort them correctly ;) well, I'm not sure if all kinds of VPN are SSL/TCP based. The author mentioned remote offices so this might be useful for him. I'd suggest to route VNC packets rather over internet than VPN (so do I), as VPN usually has the highest priority. Unless QoS is implemented packets are first come first served. There is no usually has the highest priority. Routing one over the Internet versus over the VPN won't change that priority. ok. probably I've misread somewhere about switches which QoS enabled is by default. By the way we do ask our ISP to prioritize VPN packets and they do. j On Thu, May 7, 2009 at 11:33 PM, Roberto Piola roberto.pi...@visiant.it wrote: I do not have examples, but if you are using the 1700 series router in order to originate the ipsec vpn, you may use command qos pre-classify (please search for it on cco.cisco.com) On Thu, May 7, 2009 at 9:54 PM, Brent Davidson br...@texascountrytitle.com wrote: I've got multiple satellite office all linked back to the main office via VPN. Each office has their own asterisk server which registers back to the main office's Asterisk server. Each office also has a 1Mb downstream / 384k - 768k upstream connection. The branches are using Speex for their connections back to the main office. The issue I'm having is that there are times that I need to VNC in to machines at the various offices for tech support while the user is also on the phone. Unfortunately the VNC connection apparently takes priority and makes it impossible for me to understand anything the person on the phone is saying, although they can still hear me fine. Our Main office uses a Cisco PIX 506 for the main firewall and VPN concentrator. Each branch office used a Cisco 1700 series router with IPSec enabled in the IOS. Is there any sort of QoS I can turn on on the main router or the branch routers to make sure the voice quality takes precedence over the VNC? (Any example configs would be greatly appreciated) Would I be better off routing the voice packets over the internet rather than the VPN, and could I safely do that without exposing the asterisk boxes to unnecessary security risks? (At present all of our asterisk boxes are behind the firewalls and only talk to each other over the VPN. All PSTN connection is done through TDM boards so they have no direct exposure to the internet.) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mvh, Aurimas Skirgaila ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mvh, Aurimas Skirgaila ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QoS VPN
On Thu, May 7, 2009 at 3:54 PM, Brent Davidson br...@texascountrytitle.com wrote: I've got multiple satellite office all linked back to the main office via VPN. Each office has their own asterisk server which registers back to the main office's Asterisk server. Each office also has a 1Mb downstream / 384k - 768k upstream connection. The branches are using Speex for their connections back to the main office. The issue I'm having is that there are times that I need to VNC in to machines at the various offices for tech support while the user is also on the phone. Unfortunately the VNC connection apparently takes priority and makes it impossible for me to understand anything the person on the phone is saying, although they can still hear me fine. VNC is very asymmetric. It doesn't generate much traffic from the person viewing, and it generates lots of traffic FROM the system being viewed. This helps explain why the system being viewed side can hear incoming voice packets, and outbound voice packets that have to compete with the large amount of outgoing video signal data lose. QoS may or may not help you here. If voice quality is important, you should have a separate connection dedicated to just voice. The obvious workaround is grab your cell phone and call them with that. You DO have a way to dial directly to that office without going over the PIX, right, right? How do you call the remote office when the PIX goes down? What will help you is getting a bigger line or separating the voice traffic from the data traffic completely. If you are good with ssh, you can also do a compressed ssh tunnel to encrypt and on-the-fly compress the VNC session. But if this is Windows good luck with that. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QoS VPN
I would think that VoIP over VPN is a bad idea as UDP packets need to be in realtime not corrected by the TCP of the VPN. Garth van Sittert Technical Director BitCo 08600 24826 www.bitco.co.za Aurimas Skirgaila wrote: Despite the VPN overhead, running VOIP through VPN is good idea because VPN reorders encapsulated UDP packets in correct order. Security matters as well. I'd suggest to route VNC packets rather over internet than VPN (so do I), as VPN usually has the highest priority. On Thu, May 7, 2009 at 11:33 PM, Roberto Piola roberto.pi...@visiant.it mailto:roberto.pi...@visiant.it wrote: I do not have examples, but if you are using the 1700 series router in order to originate the ipsec vpn, you may use command qos pre-classify (please search for it on cco.cisco.com http://cco.cisco.com) On Thu, May 7, 2009 at 9:54 PM, Brent Davidson br...@texascountrytitle.com mailto:br...@texascountrytitle.com wrote: I've got multiple satellite office all linked back to the main office via VPN. Each office has their own asterisk server which registers back to the main office's Asterisk server. Each office also has a 1Mb downstream / 384k - 768k upstream connection. The branches are using Speex for their connections back to the main office. The issue I'm having is that there are times that I need to VNC in to machines at the various offices for tech support while the user is also on the phone. Unfortunately the VNC connection apparently takes priority and makes it impossible for me to understand anything the person on the phone is saying, although they can still hear me fine. Our Main office uses a Cisco PIX 506 for the main firewall and VPN concentrator. Each branch office used a Cisco 1700 series router with IPSec enabled in the IOS. Is there any sort of QoS I can turn on on the main router or the branch routers to make sure the voice quality takes precedence over the VNC? (Any example configs would be greatly appreciated) Would I be better off routing the voice packets over the internet rather than the VPN, and could I safely do that without exposing the asterisk boxes to unnecessary security risks? (At present all of our asterisk boxes are behind the firewalls and only talk to each other over the VPN. All PSTN connection is done through TDM boards so they have no direct exposure to the internet.) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mvh, Aurimas Skirgaila ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QoS VPN
On Friday 08 May 2009 10:07:43 Garth van Sittert wrote: I would think that VoIP over VPN is a bad idea as UDP packets need to be in realtime not corrected by the TCP of the VPN. Not all VPNs use TCP. OpenVPN, in particular, uses UDP for the backbone. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QoS VPN
I would think that VoIP over VPN is a bad idea as UDP packets need to be in realtime not corrected by the TCP of the VPN. That depends very much on the VPN in use. OpenVPN doesn't suffer from this problem. Although it's SSL-based (and one might think it does everything through SSL-over-TCP), it actually sends the VPN traffic via UDP... it uses TCP only for the negotiation and administrative aspects of setting up the VPN connection. As far as I know, OpenVPN makes no attempt at all to re-order the packets that it encapsulates and transmits. It simply accepts the IP packets it is to carry, encrypts them individually, wraps them in UDP, and retransmits them to its peer. The peer receives the UDP, decrypts, and forwards. No re-ordering. There may be other VPNs which actually carry all of the VPN'ed data in a single TCP stream... but I think this is generally agreed to be a Bad Idea for several reasons. I run SIP over OpenVPN between my Nokia N810 handheld, and my Asterisk server at home. I have not noticed any difference in call quality between SIP-over-OpenVPN, and non-VPN'ed SIP, between these two endpoints... except, of course, when the OpenVPN-encapsulated traffic gets through, and non-VPN'ed traffic doesn't due to firewall or NATing problems at a particular wireless network access point. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QoS VPN
Dave Platt wrote: OpenVPN doesn't suffer from this problem. Although it's SSL-based (and one might think it does everything through SSL-over-TCP), it actually sends the VPN traffic via UDP... it uses TCP only for the negotiation and administrative aspects of setting up the VPN connection. UDP is the default, but OpenVPN can be configured for TCP as well ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QoS VPN
It's been a few years ago, but Network Computing had tests results showing that VoIP over a VPN was measurably better than outside a VPN. Why? Because the latency was low enough that lost UDP packets (within the VPN tunnel) could be re-transmitted before the jitter buffer had expired. Since most jitter buffers are on the order for 10 to 80 msec, if your one-way latency is any greater than a third of your jitter buffer, it's of no use. For example, if the one-way latency is 15 msec, the best-case scenario is that with single-time packet loss, the other packet would arrive at the destination in ~45 msec. Frank -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Garth van Sittert Sent: Friday, May 08, 2009 10:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] QoS VPN I would think that VoIP over VPN is a bad idea as UDP packets need to be in realtime not corrected by the TCP of the VPN. Garth van Sittert Technical Director BitCo 08600 24826 www.bitco.co.za Aurimas Skirgaila wrote: Despite the VPN overhead, running VOIP through VPN is good idea because VPN reorders encapsulated UDP packets in correct order. Security matters as well. I'd suggest to route VNC packets rather over internet than VPN (so do I), as VPN usually has the highest priority. On Thu, May 7, 2009 at 11:33 PM, Roberto Piola roberto.pi...@visiant.it mailto:roberto.pi...@visiant.it wrote: I do not have examples, but if you are using the 1700 series router in order to originate the ipsec vpn, you may use command qos pre-classify (please search for it on cco.cisco.com http://cco.cisco.com) On Thu, May 7, 2009 at 9:54 PM, Brent Davidson br...@texascountrytitle.com mailto:br...@texascountrytitle.com wrote: I've got multiple satellite office all linked back to the main office via VPN. Each office has their own asterisk server which registers back to the main office's Asterisk server. Each office also has a 1Mb downstream / 384k - 768k upstream connection. The branches are using Speex for their connections back to the main office. The issue I'm having is that there are times that I need to VNC in to machines at the various offices for tech support while the user is also on the phone. Unfortunately the VNC connection apparently takes priority and makes it impossible for me to understand anything the person on the phone is saying, although they can still hear me fine. Our Main office uses a Cisco PIX 506 for the main firewall and VPN concentrator. Each branch office used a Cisco 1700 series router with IPSec enabled in the IOS. Is there any sort of QoS I can turn on on the main router or the branch routers to make sure the voice quality takes precedence over the VNC? (Any example configs would be greatly appreciated) Would I be better off routing the voice packets over the internet rather than the VPN, and could I safely do that without exposing the asterisk boxes to unnecessary security risks? (At present all of our asterisk boxes are behind the firewalls and only talk to each other over the VPN. All PSTN connection is done through TDM boards so they have no direct exposure to the internet.) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mvh, Aurimas Skirgaila ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QoS VPN
David Backeberg wrote: On Thu, May 7, 2009 at 3:54 PM, Brent Davidson br...@texascountrytitle.com wrote: I've got multiple satellite office all linked back to the main office via VPN. Each office has their own asterisk server which registers back to the main office's Asterisk server. Each office also has a 1Mb downstream / 384k - 768k upstream connection. The branches are using Speex for their connections back to the main office. The issue I'm having is that there are times that I need to VNC in to machines at the various offices for tech support while the user is also on the phone. Unfortunately the VNC connection apparently takes priority and makes it impossible for me to understand anything the person on the phone is saying, although they can still hear me fine. VNC is very asymmetric. It doesn't generate much traffic from the person viewing, and it generates lots of traffic FROM the system being viewed. This helps explain why the system being viewed side can hear incoming voice packets, and outbound voice packets that have to compete with the large amount of outgoing video signal data lose. QoS may or may not help you here. Well, the fact that our central office has a 10mb downstream / 5mb upstream connection (Two 5Mb down 2.5Mb up DSl connections load shared) helps with them hearing me clearly too, I'm sure. I can get the packets to them faster than they can get packets to me. If voice quality is important, you should have a separate connection dedicated to just voice. The obvious workaround is grab your cell phone and call them with that. You DO have a way to dial directly to that office without going over the PIX, right, right? How do you call the remote office when the PIX goes down? What will help you is getting a bigger line or separating the voice traffic from the data traffic completely. If you are good with ssh, you can also do a compressed ssh tunnel to encrypt and on-the-fly compress the VNC session. But if this is Windows good luck with that. Yes, we can dial all satellite office through the PSTN if we really want to, but one of the reasons we went to a VOIP system was to cut down on the long-distance charges that result from office-to-office calls, and to be able to transfer calls from one office to another. All in all the system works as designed, except for the rare occasions that I'm doing support with VNC and have a person on the remote extension as well. But just because nobody else has complained yet doesn't mean there aren't other conditions that could trigger a poor-quality call. If I can find a solution that works in my worst-case VNC situation then maybe I'll prevent a few future issues from ever becoming real problems. Separating the voice off to it's own connection would defeat the cost-cutting reasoning behind the system. Thanks, Brent ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QoS VPN
Jeremy Mann wrote: Access-list 100 permit ip host asterisk server any Class-map match-any voip Match access-group 100 Policy-map voip Class voip Priority 256 Class class-default Fair-queue Interface fastethernet 0 Service-policy output voip Above is what I do to prioritize 256kbit of outbound bandwidth to voip calls, adjust accordingly. You must also use the qos pre-classify in your ipsec tunnel definitions for this to work, but it does work well. I know I'm potentially mapping other traffic than voip, but I'm lazy and don't want to classify the rtp and sip and iax ports, rarely does the box do any other traffic than voip as updates occur in off hours. You'll probably additionally want to match your ipsec keying traffic and give it priority bandwidth, if you're going to push voip through the tunnel you'll find yourself rekeying more often and want to make sure on a saturated link it gets priority so the tunnels don't drop. If you're on DSL, you probably want to research cascading the Qos, have a root policy that throttles all bandwidth to a certain speed, then a child policy that prioritizes that bandwidth, so you don't saturate your outbound circuit(think in terms of P2P protections). Thank you. This is EXACTLY what I was looking for. Do the packet counters for show policy-map int fast 0/0 only increment when the queuing kicks in or should they be incrementing all the time as packets flow? Thanks again, Brent ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] QoS VPN
I've got multiple satellite office all linked back to the main office via VPN. Each office has their own asterisk server which registers back to the main office's Asterisk server. Each office also has a 1Mb downstream / 384k - 768k upstream connection. The branches are using Speex for their connections back to the main office. The issue I'm having is that there are times that I need to VNC in to machines at the various offices for tech support while the user is also on the phone. Unfortunately the VNC connection apparently takes priority and makes it impossible for me to understand anything the person on the phone is saying, although they can still hear me fine. Our Main office uses a Cisco PIX 506 for the main firewall and VPN concentrator. Each branch office used a Cisco 1700 series router with IPSec enabled in the IOS. Is there any sort of QoS I can turn on on the main router or the branch routers to make sure the voice quality takes precedence over the VNC? (Any example configs would be greatly appreciated) Would I be better off routing the voice packets over the internet rather than the VPN, and could I safely do that without exposing the asterisk boxes to unnecessary security risks? (At present all of our asterisk boxes are behind the firewalls and only talk to each other over the VPN. All PSTN connection is done through TDM boards so they have no direct exposure to the internet.) Thanks, Brent Davidson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QoS VPN
I do not have examples, but if you are using the 1700 series router in order to originate the ipsec vpn, you may use command qos pre-classify (please search for it on cco.cisco.com) On Thu, May 7, 2009 at 9:54 PM, Brent Davidson br...@texascountrytitle.comwrote: I've got multiple satellite office all linked back to the main office via VPN. Each office has their own asterisk server which registers back to the main office's Asterisk server. Each office also has a 1Mb downstream / 384k - 768k upstream connection. The branches are using Speex for their connections back to the main office. The issue I'm having is that there are times that I need to VNC in to machines at the various offices for tech support while the user is also on the phone. Unfortunately the VNC connection apparently takes priority and makes it impossible for me to understand anything the person on the phone is saying, although they can still hear me fine. Our Main office uses a Cisco PIX 506 for the main firewall and VPN concentrator. Each branch office used a Cisco 1700 series router with IPSec enabled in the IOS. Is there any sort of QoS I can turn on on the main router or the branch routers to make sure the voice quality takes precedence over the VNC? (Any example configs would be greatly appreciated) Would I be better off routing the voice packets over the internet rather than the VPN, and could I safely do that without exposing the asterisk boxes to unnecessary security risks? (At present all of our asterisk boxes are behind the firewalls and only talk to each other over the VPN. All PSTN connection is done through TDM boards so they have no direct exposure to the internet.) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QoS VoIP
A clearer explanation of your problem, including examples and output, is needed. Anael DIAZ wrote: Hi! I have some problem in my asterisk 1.4.2, I've installed it on centOS 5.2 and this didn't accept voip QoS and can't route the packets having voip QoS. So I should change voip packets to be routing with centOS. I want to use iproute2 but i don't what to do after installing iproute2. Anyone could help me please? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QoS VoIP
Anael DIAZ wrote: Hi! I have some problem in my asterisk 1.4.2, I've installed it on centOS 5.2 and this didn't accept voip QoS and can't route the packets having voip QoS. So I should change voip packets to be routing with centOS. I want to use iproute2 but i don't what to do after installing iproute2. Anyone could help me please? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This stuff is pure voodoo. I've found very little good specific instruction. I put this into rc.local to set up QoS. I'm not sure I understood it then, and I'm sure I don't understand it now, but it may be useful to you. I also put the various tos stuff in sip.conf, etc. cat tos.local ## eth1 is the external interface ## remove the queues EXTIF=eth1 tc qdisc del dev $EXTIF root ## This is to set up QoS for voip - specifically iax. ## from http://www.howtoforge.com/voip_qos_traffic_shaping_iproute2_asterisk ## ethx is the *external* port tc qdisc add dev $EXTIF root handle 1: prio priomap 2 2 2 2 2 2 2 2 1 1 1 1 1 1 1 0 tc filter add dev $EXTIF protocol ip parent 1: prio 1 u32 match ip dport 4569 0x flowid 1:1 tc filter add dev $EXTIF protocol ip parent 1: prio 1 u32 match ip sport 4569 0x flowid 1:1 tc filter add dev $EXTIF protocol ip parent 1: prio 1 u32 match ip tos 0x10 0xff flowid 1:2 Please post anything you do find. Good luck. sean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] QOS and Asterisk
I will have a small shop with ~4 phones using an HP server with Asterisk on it, it has two NICS and so I planned on plugging one into the cable modem, and the other into the switch. I was going to let this box perform NAT for the company but I am concerned about QOS for the VOIP portion. Anyone got a similar setup and care to share what they successfully implemented? Thanks! jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QOS and Asterisk
On Thu, 15 May 2008 15:39:26 -0600, Joseph L. Casale wrote: I will have a small shop with ~4 phones using an HP server with Asterisk on it, it has two NICS and so I planned on plugging one into the cable modem, and the other into the switch. I was going to let this box perform NAT for the company but I am concerned about QOS for the VOIP portion. Anyone got a similar setup and care to share what they successfully implemented? Thanks! jlc You should take a serious look at Astlinux. It's en embedded Asterisk distro that handles routing, including QoS, when necessary. See www.astlinux.org. Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QOS and Asterisk
You SHOULD be concerned with QOS. All the way to an including the vendor or your service cold really sucku Michael Graves wrote: On Thu, 15 May 2008 15:39:26 -0600, Joseph L. Casale wrote: I will have a small shop with ~4 phones using an HP server with Asterisk on it, it has two NICS and so I planned on plugging one into the cable modem, and the other into the switch. I was going to let this box perform NAT for the company but I am concerned about QOS for the VOIP portion. Anyone got a similar setup and care to share what they successfully implemented? Thanks! jlc You should take a serious look at Astlinux. It's en embedded Asterisk distro that handles routing, including QoS, when necessary. See www.astlinux.org. Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QOS for outgoing SIP calls
Chris Mason wrote: QOS can only be on outgoing, you can't set the priority of a packet after you receive it. The only other solution would be the cooperation of the ISP to provide QOS upstream of you. Good luck. QOS is probably not the most precise term as it's normally associated with RSVP, MPLS, packet headers, etc. But you can, in Netscreens at least, define a Guaranteed Bandwidth. We do this for SIP/IAX IPs, in both outgoing and incoming policies, and it works both ways. Audio quality is good and there are no chan_sip.c: Peer is now (UNREACHABLE|Lagged) messages even during long DVD or Bitorrent xfers. The reason it works outbound is a no-brainer, but inbound bandwidth is also effectively guaranteed. Sure there's no way to control external devices that ignore ICMP source-quench or break TCP congestion control but those flows are typically limited to nefarious sources which would not be responsive to other types of QOS anyhow (BGP being one potential exception). Roger Marquis ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QOS for outgoing SIP calls
Simon wrote: Hi There, We have our Asterisk box using a external SIP provider for outgoing calls over our DSL line. This seems to be going well... But i do have the ability to set some QOS ports in our linksystem DSL router... Its faily basic, so im wondering if it will help at all... We can specify High, Med, Low settings for: FTP, HTTP, Telnet, SMTP and POP3. Plus we have the ability to specify up to 3 ports for the same settings. Is this worth doing? If so, what ports should i specifiy? Simon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users If you can, try giving the highest priority to the UDP protocol or the provider IP address. Sam ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QOS for outgoing SIP ... Who needs QoS anyway!
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Simon wrote: | Is this worth doing? If so, what ports should i specifiy? http://www.bricklin.com/qos.htm -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iQIVAwUBSAcxA4OeOV2sx4+mAQKrKg//Vs5n0K1m1+C+sTSol2a1MbFHU/QCh1fT u5IwLOvQwcDvxOHikYYk8Ornm5FEbp4cewnCVKS9BeLkurlaQ1qXiwPdCHhnrNhn q2ufIskPudXn2cNj9pbAkZhmNL7R7m1XruITBGrXvV4d2WAZy6lmNGnVOhipU5ff HIV1aAacawCJG6oJjpD/NvjydHYwP6KgvOJkcLICrb7FEC4bfDfULQxThFF9Jzf3 aMzvddM+GdBGzhT0q7FH7JGQmuahlN1kdIyLY8Rw+/ouEgm4xYeyZ486JaBk2xOK 5ZnYqXXLmNuB7LPIkSCp2Fi5usFUBrKq8nanbvonw9Te3pILCG/FAfg/+O3Y2ZQb 0aZsUW13BHj9hfiZYLnKGCeJV1hLLuLWH+fP7E9kzFbi8ls7/Ke+oe7l8fgRFfzt oaInPjl1tshzbaOesSX1H8OI5QyfGgmuhyVu5E0tFmy9HX7QnxxBrI/GXMwQ3F6/ +Qv058sJ5qjQtGMi0fI6GoDa3xQRCzyBgWjuOBHhk64FVnMs3Rdoti69YhsWH52a LQMleyChhVQ0nrP5eVaykEryLNRw7jDV1X/ivtPNOHfQ0fN8337AHPmmKpzo22sK xdzK+RY1I/qZ+SOD6YPaiKjxaB9gqDPe7jGy41NlGsjnrgUjJh2c2/tvcyDYaW4u 0J5kiHsMXLI= =oY+k -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QOS for outgoing SIP ... Who needs QoS anyway!
On Thu, 2008-04-17 at 07:16 -0400, sil wrote: Simon wrote: | Is this worth doing? If so, what ports should i specifiy? http://www.bricklin.com/qos.htm Yeah, well, that's all fine and dandy as long as more capacity is an option. Many people are already subscribed to the most capacity available to them and using it. b. signature.asc Description: This is a digitally signed message part ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QOS for outgoing SIP ... Who needs QoS anyway!
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Brian J. Murrell wrote: | | Yeah, well, that's all fine and dandy as long as more capacity is an | option. Many people are already subscribed to the most capacity | available to them and using it. | | b. Apparently man people don't understand that those QoS settings on routers mean little most of the time. Most providers resell QoS as a premium service, so while many waste their time painting their packets those markings get stripped. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iQIVAwUBSAc5R4OeOV2sx4+mAQKEpQ//dYu+9MFaHgHzbBntTMbUHuY4usW5Aq+L crMlq3nYqgi8kWfVShhEozKHvtaYc7J7YBSkE2QprhM/YTp+wE3Oy9NM5GU6Ckhz IDaFNteO62zyxg5ljE81iIQd0tTJjutIf3FQVZBegzpINGIiEkjKBfbx/4UiO6HL bexoS3pnV4xjjS8xO8rMNl8+1XVubpG42K1/alw0G7y7W9Pog+u67+dLx1Tnx0EX RTlAeLZ64u5hy7CXeRdLSM3Onn8IuCnOIP2Py4OEUjLH8K4yMb83IVlhv+KSp4q4 5Tw7LWFsM/NZ0J6xz3MeUnXJHOkNK6Z5UJAfV1LmjiWdpxDCfYDifu6Y5D425+po gd/zHRI+SZJAhzN4l0oWIxSRQdCL6APyFqYFftO9bxAzDoK6EMXADIPvc3Ovb/A0 eUh6rZAe3y5/FfQy29GN23u5//ahFDCzQ9YqhbDjLEc/Z+PLi/lsEdWwWMrUMyus Q4nBs9osuxjRZYWEKUTLal+ItNL/BSiqHurN1T/l3W1/xigYiZHByxEBI2/+jYX6 66wQU6CSE2YC+n9R+rbsAP5OawOTxpXnDdTXEydHCPgdOAS5HmrwTp0t5MNZ4V/N iSGIBBAcV0HJIKRKaeGweIRGStAQPXbfQ9Qha7uYOqnyYwPbt18/vw08YlbdXXCO woCJ+I+AchI= =+Lgr -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QOS for outgoing SIP ... Who needs QoS anyway!
On Thu, 2008-04-17 at 07:54 -0400, sil wrote: Apparently man people don't understand that those QoS settings on routers mean little most of the time. Most providers resell QoS as a premium service, so while many waste their time painting their packets those markings get stripped. Maybe your understanding of QOS and mine is different. Of course I have no illusions that I can assign a priority to my packets that is going to be meaningful to anyone once they leave my network. But certainly at my choke point which is of course my Internet uplink, I can apply QOS (i.e. traffic shaping, which is what the OP's router was offering) to make sure that what little capacity is there is giving priority to my voice traffic. Think of my ISP uplink as that moderately congested road in which emergency vehicles need to have other casual traffic pull over and let it through. Traffic shaping is the effect of those vehicles pulling over and letting the voice traffic through in priority. This is exactly what OP's router was allowing him to do, albeit in what sounds like a really crappy way -- only 3 ports or something like that. b. signature.asc Description: This is a digitally signed message part ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QOS for outgoing SIP ... Who needs QoS anyway!
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Brian J. Murrell wrote: | Maybe your understanding of QOS and mine is different. Of course I have | no illusions that I can assign a priority to my packets that is going to | be meaningful to anyone once they leave my network. | | But certainly at my choke point which is of course my Internet uplink, I | can apply QOS (i.e. traffic shaping, which is what the OP's router was | offering) to make sure that what little capacity is there is giving | priority to my voice traffic. | | Think of my ISP uplink as that moderately congested road in which | emergency vehicles need to have other casual traffic pull over and let | it through. Traffic shaping is the effect of those vehicles pulling | over and letting the voice traffic through in priority. This is exactly | what OP's router was allowing him to do, albeit in what sounds like a | really crappy way -- only 3 ports or something like that. | | b. Let's take a bare bones look at this. Let's say your connection is 300k and you have five packets coming in at 60k each to saturate your network: Provider to you Packet 1 You Packet 2 You Packet 3 You Packet 4 You Packet 5 You You believe that this is happening: Packet 1 You --- This is voice send it first -- Device Packet 2 You --- This is voice send it first -- Device Packet 3 You --- This is P2P leave it 4 last -- Device Packet 4 You --- This is P2P leave it 4 last -- Device Packet 5 You --- This is AIM make it second! -- Device Its fine and dandy, but the problem is you're still getting 5 packets. You're still saturated period. No QoS in the world outside of your provider and more bandwidth can alleviate that. Your provider is not going to care what you do once its passed to the CPE. So look at it logically again. QoS on a home router... Useless COMING IN. Going out... Means little but helps MINIMALLY. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iQIVAwUBSAdCdoOeOV2sx4+mAQImCA//Q3pXHy/hUUqc/RvN/WNXzYiXMVR5gmKR xdNc+GZkN0ks16wKqJLxXITDwXE9vWygEmY3G97xo9f3jFR/NihtiTDTo7n/nvA6 GDC1gOw5UY20793ACHdL4mroCL7A8UMUdZGDZyyhQVSIpKZ4Uhk9bwgDPzRCYkOp rwgL2WyQAPrk6GjKg/XIg3H6vBtI6ZcuXV5xu5CoPxOb1hPEzj/AX/OiQbZlAGPY CLFWnVSs1YBM4rq2Jt3KA7kKPsFST81JMMWSxU+axKzmaa6LmU29FgX4WG8jBG5s 0Nxk0PkXIzu6XfLVkU8Dop5FCUpxbDRmh6OyXyvluQ2SEBh48ZiPSnClDI+Ue9JN J5z2QQen8qtK/HdbCDp08MF6MSiEceYYCcwWHGMg9KlD3u2FgY9rrPZ3hKrP9Tz5 1ciLXig5mvyhWBGIS5mIhg7QnnWAzMsXjbQ8buHgir82ptDbM3wSyWdkWHNR47Fr uFe+QGVV4JHFzHsDkeo/qfGA2juwazMfNXJyV67vWnyZNhnhtZ+kEAbMXeABvhjQ rw/bgtq7gdiv/fwIgq51WKEPQbyHozRpqdyZPUzBJBsDND5iKivzIboug+hS4QJH tqK/c29mir/0D5CbXswhCTjbiUIYIyH8Gu+OU3G1uhNrv+TRm7E+8jCtvM+zfvXu AaTY8D7BmJo= =ldzm -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QOS for outgoing SIP ... Who needs QoS anyway!
On Thu, 2008-04-17 at 08:36 -0400, J. Oquendo wrote: Brian J. Murrell wrote: | But certainly at my choke point which is of course my Internet uplink, ^^^ I | can apply QOS (i.e. traffic shaping, which is what the OP's router was | offering) to make sure that what little capacity is there is giving | priority to my voice traffic. Let's take a bare bones look at this. Let's say your connection is 300k Downstream or upstream? Notice I said Internet uplink in my previous message. Anyone at all familiar with traffic shaping understands that they can only shape the uplink, not the downlink. The best you can do with the downlink is to police it to try to keep the congestion below 100%. But that's mostly alright given how the ISPs have perverted the Internet with asymmetric last mile connections to consumers. and you have five packets coming in at 60k each to saturate your network: First of all,your whole example is pointless as you are clearly talking about downstream and I have already said that anyone knowledgeable with traffic shaping knows you cannot shape the downlink only the uplink. However, let's see where else your example fails. My MTU is only about 1500 bytes or so, so 60k packets to me are impossible. I'd tend to guess that for most of the Internet, packets max out at about 1500 given the prevalence of ethernet connected devices. So in order to saturate my 300k you'd have to send me 200 packets all in that one second. Provider to you Packet 1 You Packet 2 You Packet 3 You Packet 4 You Packet 5 You You believe that this is happening: Packet 1 You --- This is voice send it first -- Device Packet 2 You --- This is voice send it first -- Device Packet 3 You --- This is P2P leave it 4 last -- Device Packet 4 You --- This is P2P leave it 4 last -- Device Packet 5 You --- This is AIM make it second! -- Device As I've said, you cannot shape this traffic. I've already conceded that. But again, OP was talking about uplink shaping, not downlink. Its fine and dandy, but the problem is you're still getting 5 packets. You're still saturated period. Right. You cannot shape the downlink. You can only police it to prevent packet loss. No QoS in the world outside of your provider and more bandwidth can alleviate that. If more bandwidth is an option, but I already stated that for many people, it's not an option. They have exactly one or two choices and they are subscribed to their maximum available. QoS on a home router... Useless COMING IN. Going out... Means little but helps MINIMALLY. Not at all little. If you have a lot of low priority outgoing traffic (i.e. p2p) saturating your link, uplink traffic shaping will mean the difference between a completely unintelligible call and something very acceptable. b. signature.asc Description: This is a digitally signed message part ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QOS for outgoing SIP ... Who needs QoS anyway!
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Brian J. Murrell wrote: | Not at all little. If you have a lot of low priority outgoing traffic | (i.e. p2p) saturating your link, uplink traffic shaping will mean the | difference between a completely unintelligible call and something very | acceptable. Is it? So you're telling me if you're saturated on the way in, fixing up your packets on the way out is the solution. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iQIVAwUBSAdPUoOeOV2sx4+mAQJT6w/7Bm4hcyAaaLlwlYo8Dsfw5oUCUOW+0TLy jrIWS1piOYbe+MBjpoliOF7nJETrQnFUP5y/kZjjOTxCuyz1XLlj7ExOdddGXudp My2tO81/Gkx/kicOQDtdxLKtMBWI8ix3Ef1Z1dtQIh4DYJqSGbzgTez5D1WfxhXG IZ2IFq8CKzpjT7oAExmo/l7QqetCXPMgM4gZ24CDXlESL/esYBJL5sWfzG+4dLyJ 4INMpPnckXjdf/WyCIeMDrGRAEKpNQ8Ls+X/EAgwqJ83Z6iTJUrW6xMfO9KXAlDP BNwrX1/Xlx0quNd+tH+u0j8DcQ0sy9jt4KixOQYqCb9VtpDz5Ucf8zyqMC277ugz FwaDSpUlkASe/JK0m/IFf4lvnrgBna1jDFa5k13u8R+Ja1rcb0+S7I5Rk6MxpBCo xIfRIGHqO/hmAv3ckj2qIoGetlPZNTT94fgGV/d5UnAU4eOTuNXeZURS9Wf3XVN6 Yc90oGHKWfB3O0XJNS/QI4LeI7BxWJUDmyC1PczKfhIj9ox9K+GD1tSvto3nSqZE NFpdcG7Ch1EDAYZuptvAp+3tKy+ifLYmultAq7/ehBeJ+t0GJxxwFqIYTGwuUCBh M0Gd570V4baOhl3UY917uwkTb4bBXS+9wh2J7qTUqVmGlOYC/6x0MrJLabetQiKT A5+VziQWa/Y= =tSIZ -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QOS for outgoing SIP ... Who needs QoS anyway!
On Thu, 2008-04-17 at 09:25 -0400, J. Oquendo wrote: Is it? So you're telling me if you're saturated on the way in, fixing up your packets on the way out is the solution. I think I've made it clear that my argument is only about uplink shaping and the requirement for it given the asymmetric nature of a lot of last mile connections existing today. Funny enough that is *exactly* what the OP was asking about. b. signature.asc Description: This is a digitally signed message part ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QOS for outgoing SIP ... Who needs QoS anyway!
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Brian J. Murrell wrote: | I think I've made it clear that my argument is only about uplink shaping | and the requirement for it given the asymmetric nature of a lot of last | mile connections existing today. Funny enough that is *exactly* what | the OP was asking about. | | b. Answers the question with minimal relevance, not even a band-aid solution. You fixing up inbound traffic will do nothing for a horrible conversation if you're congested coming in. Solution would be to add more bandwidth. Else you could fiddle around around creating all the fuzzy rules on the planet shaping traffic all sorts of methods once its in your CPE but this WILL NOT HELP YOU HAVE A BETTER CONVERSATION. When it does, when someone can realistically point this out please let me know so I can switch from a DS3 to T1 and save money. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iQIVAwUBSAdXC4OeOV2sx4+mAQIVFw/+NttSjlj132/ikQZN4pI6kJJH49GxJiMp aA9ugBu0jA9ZXSgU8oHw9ZbgkZfjalM1vtmekOW+w4eXUwlx82jEGEJ1e7iBT30e wB9cOOMlpn1+sFrZxesAz/8a7ziFC02Ydf2+V3j8FfPga8DuHWtF/hubm7p/d3Zy Km1Vm1ruajCTM9PAvVO/Jj2TybzYwWj7Pj2TzwZEsYCXUuj5/E0fnQjJKCI4q6e6 UHsOs5tpqedzRCSJ2Zv96xkHAFWDLOUke2vXp20ZETnOxqOVtULm+EuYXsHvauYN 6sMZf7Tq04+jMrbR1GWLCevvEoJN1XpTEOBb3yv7S/7U7Ih/mQfluHNj4hVUACYs vFlIJyHBeLxeAOH5VFm66SDtIQ2TKGLuFblDD5E6MmhXYdhwdwsmGecfaEJHR/+K 83CDQ1P1tDtN6JjcYXsoN8125uRKYH2EQunfZq01GJQlj6QNJcZHcv9FrRYXan42 7yxB+h1UpgNLMAthOQsQ8+nt7rRD8v0GlPZBwXlRF1n2S2jAVJiwlrihfiW5xA6C LsRuU7GIo/XkX/zNQk2BIGszziIEGcYaJjYnXBdsP2QN6IwkCz8xwQbgtssSMqmd kbFelVI4BepzbG2lUVkmUAavoFL7T1c9eIyMU9vunOJtP/azTadXP9ITS936mKYK pYOvun1cnqU= =I0yU -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QOS for outgoing SIP ... Who needs QoS anyway!
May I suggest the following read: A Beginners Guide To Successful VOIP Over DSL http://www.smallnetbuilder.com/content/view/30340/83/ Which covers both QoS and traffic shaping in small routers. It was written based upon my own experience with both Asterisk and hosted PBX providers. Michael On Thu, 17 Apr 2008 07:16:40 -0400, sil wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Simon wrote: | Is this worth doing? If so, what ports should i specifiy? http://www.bricklin.com/qos.htm -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iQIVAwUBSAcxA4OeOV2sx4+mAQKrKg//Vs5n0K1m1+C+sTSol2a1MbFHU/QCh1fT u5IwLOvQwcDvxOHikYYk8Ornm5FEbp4cewnCVKS9BeLkurlaQ1qXiwPdCHhnrNhn q2ufIskPudXn2cNj9pbAkZhmNL7R7m1XruITBGrXvV4d2WAZy6lmNGnVOhipU5ff HIV1aAacawCJG6oJjpD/NvjydHYwP6KgvOJkcLICrb7FEC4bfDfULQxThFF9Jzf3 aMzvddM+GdBGzhT0q7FH7JGQmuahlN1kdIyLY8Rw+/ouEgm4xYeyZ486JaBk2xOK 5ZnYqXXLmNuB7LPIkSCp2Fi5usFUBrKq8nanbvonw9Te3pILCG/FAfg/+O3Y2ZQb 0aZsUW13BHj9hfiZYLnKGCeJV1hLLuLWH+fP7E9kzFbi8ls7/Ke+oe7l8fgRFfzt oaInPjl1tshzbaOesSX1H8OI5QyfGgmuhyVu5E0tFmy9HX7QnxxBrI/GXMwQ3F6/ +Qv058sJ5qjQtGMi0fI6GoDa3xQRCzyBgWjuOBHhk64FVnMs3Rdoti69YhsWH52a LQMleyChhVQ0nrP5eVaykEryLNRw7jDV1X/ivtPNOHfQ0fN8337AHPmmKpzo22sK xdzK+RY1I/qZ+SOD6YPaiKjxaB9gqDPe7jGy41NlGsjnrgUjJh2c2/tvcyDYaW4u 0J5kiHsMXLI= =oY+k -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QOS for outgoing SIP ... Who needs QoS anyway!
My personnal experience is if you`re looking for an inexpensive solution (SOHO), StreamEngine based routers (a lot of D-Link products are Streamengine based, for example the DI-724GU and the DIR-655) do a decent job of providing QoS on the upstream, which is where you (usually) need it anyways. And the good thing is you often do not have to do anything but set the upload bandwidth (yes there is an automatic mode, but it's not that great). Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Graves Sent: Thursday, April 17, 2008 10:34 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] QOS for outgoing SIP ... Who needs QoS anyway! May I suggest the following read: A Beginners Guide To Successful VOIP Over DSL http://www.smallnetbuilder.com/content/view/30340/83/ Which covers both QoS and traffic shaping in small routers. It was written based upon my own experience with both Asterisk and hosted PBX providers. Michael On Thu, 17 Apr 2008 07:16:40 -0400, sil wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Simon wrote: | Is this worth doing? If so, what ports should i specifiy? http://www.bricklin.com/qos.htm -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iQIVAwUBSAcxA4OeOV2sx4+mAQKrKg//Vs5n0K1m1+C+sTSol2a1MbFHU/QCh1fT u5IwLOvQwcDvxOHikYYk8Ornm5FEbp4cewnCVKS9BeLkurlaQ1qXiwPdCHhnrNhn q2ufIskPudXn2cNj9pbAkZhmNL7R7m1XruITBGrXvV4d2WAZy6lmNGnVOhipU5ff HIV1aAacawCJG6oJjpD/NvjydHYwP6KgvOJkcLICrb7FEC4bfDfULQxThFF9Jzf3 aMzvddM+GdBGzhT0q7FH7JGQmuahlN1kdIyLY8Rw+/ouEgm4xYeyZ486JaBk2xOK 5ZnYqXXLmNuB7LPIkSCp2Fi5usFUBrKq8nanbvonw9Te3pILCG/FAfg/+O3Y2ZQb 0aZsUW13BHj9hfiZYLnKGCeJV1hLLuLWH+fP7E9kzFbi8ls7/Ke+oe7l8fgRFfzt oaInPjl1tshzbaOesSX1H8OI5QyfGgmuhyVu5E0tFmy9HX7QnxxBrI/GXMwQ3F6/ +Qv058sJ5qjQtGMi0fI6GoDa3xQRCzyBgWjuOBHhk64FVnMs3Rdoti69YhsWH52a LQMleyChhVQ0nrP5eVaykEryLNRw7jDV1X/ivtPNOHfQ0fN8337AHPmmKpzo22sK xdzK+RY1I/qZ+SOD6YPaiKjxaB9gqDPe7jGy41NlGsjnrgUjJh2c2/tvcyDYaW4u 0J5kiHsMXLI= =oY+k -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QOS for outgoing SIP ... Who needs QoS anyway!
Mike wrote: do a decent job of providing QoS on the upstream, which is where you (usually) need it anyways. QOS can only be on outgoing, you can't set the priority of a packet after you receive it. The only other solution would be the cooperation of the ISP to provide QOS upstream of you. Good luck. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QOS for outgoing SIP ... Who needs QoS anyway!
On Thu, 2008-04-17 at 11:40 -0400, Chris Mason (Lists) wrote: Mike wrote: do a decent job of providing QoS on the upstream, which is where you (usually) need it anyways. QOS can only be on outgoing, Which is what he meant when he said upstream I believe. you can't set the priority of a packet after you receive it. Indeed. The only other solution would be the cooperation of the ISP to provide QOS upstream of you. Good luck. Heh. Yeah, no doubt. b. signature.asc Description: This is a digitally signed message part ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QOS for outgoing SIP ... Who needs QoS anyway!
At 05:59 AM 4/17/2008, you wrote: Not at all little. If you have a lot of low priority outgoing traffic (i.e. p2p) saturating your link, uplink traffic shaping will mean the difference between a completely unintelligible call and something very acceptable. My network looks like this: Cable modem Linksys WRT54GS running Sveasoft LAN port 1 to the phone system running on it's own set of wires LAN ports 2-4 to everything else I've set the priority on port one to the highest and the priority on all the other ports to low and as far as I can tell, we've never had an issue where a big upload has impacted our voice calls. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QOS for outgoing SIP ... Who needs QoS anyway!
J. Oquendo wrote: Its fine and dandy, but the problem is you're still getting 5 packets. You're still saturated period. No QoS in the world outside of your provider and more bandwidth can alleviate that. Your provider is not going to care what you do once its passed to the CPE. So look at it logically again. QoS on a home router... Useless COMING IN. Going out... Means little but helps MINIMALLY. I think the road to success, when talking about upstream at least, is partially paved by trying to keep maximum traffic at 4 packets instead of 5, if 5 is going to saturate the link. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QOS for outgoing SIP ... Who needs QoS anyway!
J. Oquendo wrote: it does, when someone can realistically point this out please let me know so I can switch from a DS3 to T1 and save money. Use the T1 for voice and get a DSL modem for your data use? :) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] QOS for outgoing SIP calls
Hi There, We have our Asterisk box using a external SIP provider for outgoing calls over our DSL line. This seems to be going well... But i do have the ability to set some QOS ports in our linksystem DSL router... Its faily basic, so im wondering if it will help at all... We can specify High, Med, Low settings for: FTP, HTTP, Telnet, SMTP and POP3. Plus we have the ability to specify up to 3 ports for the same settings. Is this worth doing? If so, what ports should i specifiy? Simon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QOS for outgoing SIP calls
On Wed, Apr 16, 2008 at 11:49 PM, Simon [EMAIL PROTECTED] wrote: Hi There, We have our Asterisk box using a external SIP provider for outgoing calls over our DSL line. This seems to be going well... But i do have the ability to set some QOS ports in our linksystem DSL router... Its faily basic, so im wondering if it will help at all... We can specify High, Med, Low settings for: FTP, HTTP, Telnet, SMTP and POP3. Plus we have the ability to specify up to 3 ports for the same settings. Is this worth doing? If so, what ports should i specifiy? Hi Simon, You won't be able to get much use of your router's QoS if it can only set it via port number. By default Asterisk will select a UDP port somewhere in the range of 10,000 to 20,000 to carry the RTP. The port selected for the RTP will be different at your end and at your providers end which means you would need two QoS port rules per call. You can change the port range your Asterisk server uses for RTP in rtp.conf but there's probably not a lot of point given you can't prioritise a big enough range with only 3 rules available. To be of any practical use for SIP calls you really need to be able to set QoS by IP address. Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] QoS Monitor
I would like to be able to monitor my QoS.. I see that Qwest is using this QoS Manager (Firehunter) http://www.home.agilent.com/cgi-bin/pub/agilent/Product/cp_Product.jsp?NAV_ID=-536885714.536882909.00LANGUAGE_CODE=engCONTENT_KEY=49888ID=49888COUNTRY_CODE=US I have some buddies who work at Qwest and use this software, however they are monitoring primarly Sonus GSX switches with it, has anyone used this in an asterisk environment? -=Linsys=- IntrusionSec.com #1 Hacker Gamez Web Site On the Internet http://www.intrusionsec.com [EMAIL PROTECTED] - When Your Life Flashes Before Your Eyes When You Die, Does That Include The Part Where Your Life Flashes Before Your Eyes? - ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] QoS General Question
How much of an impact can/does local network traffic have on call quality? Would opening large files on local servers affect call quality? We are running QoS on the router but that will only prioritize traffic in/out of the network. ___ 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 General Question
On Tue, 9 Aug 2005 12:07:07 -0400, Geoff Manning wrote: How much of an impact can/does local network traffic have on call quality? Would opening large files on local servers affect call quality? We are running QoS on the router but that will only prioritize traffic in/out of the network. Sure it can. If you have a network segment that's fully saturated and you're also pushing VOIP data over that segment you'll have problems. In practice most networks are not that busy, but it can happen. If your phones, switch and NICs are VLAN capable you can setup a dedicated VLAN for the voice traffic and ensure that it gets priority. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 fwd 54245 ___ 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 General Question
Michael Graves wrote: Sure it can. If you have a network segment that's fully saturated and you're also pushing VOIP data over that segment you'll have problems. In practice most networks are not that busy, but it can happen. If your phones, switch and NICs are VLAN capable you can setup a dedicated VLAN for the voice traffic and ensure that it gets priority. Michael Thanks for the info. We are experiencing issues with quality and I'm trying to smooth them out. Is there a way to determine the impact that is being caused by the local traffic? Monitoring tools that will show this in report form or realtime? Every day or so we get reports that there is a lot of problems for short bursts of time. I would like to be able to show that the local traffic is affecting this. Thanks, Geoff ___ 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 General Question
Geoff Manning wrote: Michael Graves wrote: Sure it can. If you have a network segment that's fully saturated and you're also pushing VOIP data over that segment you'll have problems. In practice most networks are not that busy, but it can happen. If your phones, switch and NICs are VLAN capable you can setup a dedicated VLAN for the voice traffic and ensure that it gets priority. Michael Thanks for the info. We are experiencing issues with quality and I'm trying to smooth them out. Is there a way to determine the impact that is being caused by the local traffic? Monitoring tools that will show this in report form or realtime? Every day or so we get reports that there is a lot of problems for short bursts of time. I would like to be able to show that the local traffic is affecting this. In my experience, for local LAN audio issues, duplex problems are the problem, not LAN traffic. Of course, if you are running Asterisk on your file server or something silly like that, all bets are off. ___ 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 General Question
On Tue, 09 Aug 2005 11:26:11 -0500, Eric Wieling aka ManxPower wrote: Geoff Manning wrote: Michael Graves wrote: Sure it can. If you have a network segment that's fully saturated and you're also pushing VOIP data over that segment you'll have problems. In practice most networks are not that busy, but it can happen. If your phones, switch and NICs are VLAN capable you can setup a dedicated VLAN for the voice traffic and ensure that it gets priority. Michael Thanks for the info. We are experiencing issues with quality and I'm trying to smooth them out. Is there a way to determine the impact that is being caused by the local traffic? Monitoring tools that will show this in report form or realtime? Every day or so we get reports that there is a lot of problems for short bursts of time. I would like to be able to show that the local traffic is affecting this. In my experience, for local LAN audio issues, duplex problems are the problem, not LAN traffic. Of course, if you are running Asterisk on your file server or something silly like that, all bets are off. Oh, yes! That's a good possibility as well, expecially with some Cisco gear. One problem that I had was related to saturating a segment during an automated backup procedure. When a server in the UK started its backup processes at an apparently idel time callers in the US had issues. What's after hours there is middle of the day over here. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 fwd 54245 ___ 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 General Question
Michael Graves wrote: Oh, yes! That's a good possibility as well, expecially with some Cisco gear. One problem that I had was related to saturating a segment during an automated backup procedure. When a server in the UK started its backup processes at an apparently idel time callers in the US had issues. What's after hours there is middle of the day over here. Michael This is a dedicated Asterisk server fortunately! So I am not competeing with anything else for network resources on the same server. Thanks, Geoff ___ 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 General Question
On Tue, 09 Aug 2005 11:26:11 -0500, Eric Wieling aka ManxPower wrote: Geoff Manning wrote: Michael Graves wrote: Sure it can. If you have a network segment that's fully saturated and you're also pushing VOIP data over that segment you'll have problems. In practice most networks are not that busy, but it can happen. If your phones, switch and NICs are VLAN capable you can setup a dedicated VLAN for the voice traffic and ensure that it gets priority. Michael Thanks for the info. We are experiencing issues with quality and I'm trying to smooth them out. Is there a way to determine the impact that is being caused by the local traffic? Monitoring tools that will show this in report form or realtime? Every day or so we get reports that there is a lot of problems for short bursts of time. I would like to be able to show that the local traffic is affecting this. In my experience, for local LAN audio issues, duplex problems are the problem, not LAN traffic. Of course, if you are running Asterisk on your file server or something silly like that, all bets are off. If this wasn't already obvious to everyone, especially newbies, this means that it is imperative to connect your network using switches, not hubs. Tom ___ 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 General Question
Geoff Manning wrote: Michael Graves wrote: Oh, yes! That's a good possibility as well, expecially with some Cisco gear. One problem that I had was related to saturating a segment during an automated backup procedure. When a server in the UK started its backup processes at an apparently idel time callers in the US had issues. What's after hours there is middle of the day over here. Michael This is a dedicated Asterisk server fortunately! So I am not competeing with anything else for network resources on the same server. Are your phones on shared links to the switch? i.e. PC - Phone - Switch? -- Eric Wieling * BTEL Consulting * 504-210-3699 x2120 ___ 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 General Question
Eric Wieling aka ManxPower wrote: Are your phones on shared links to the switch? i.e. PC - Phone - Switch? Actually it is a legacy PBX - Asterisk integration Legacy Handset -- Mitel SX 200 -- Asterisk -- Switch -- Router The calls come inbound over the internet as SIP to Asterisk and are routed into the Mitels ACD queue system where the user picks it up. Thanks, Geoff ___ 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 General Question
Eric Wieling aka ManxPower wrote: In my experience, for local LAN audio issues, duplex problems are the problem, not LAN traffic. Rock on! I am in half duplex mode: serv01:~# ethtool eth0 Settings for eth0: Supported ports: [ MII ] Supported link modes: 10baseT/Half 10baseT/Full 100baseT/Half 100baseT/Full 1000baseT/Half 1000baseT/Full Supports auto-negotiation: Yes Advertised link modes: 10baseT/Half 10baseT/Full 100baseT/Half 100baseT/Full 1000baseT/Half 1000baseT/Full Advertised auto-negotiation: Yes Speed: 100Mb/s Duplex: Half Port: Twisted Pair PHYAD: 1 Transceiver: internal Auto-negotiation: on Supports Wake-on: g Wake-on: d Current message level: 0x00ff (255) Link detected: yes This could help solve a lot of quality issues. Thanks! ___ 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 General Question
Geoff Manning wrote: Eric Wieling aka ManxPower wrote: Are your phones on shared links to the switch? i.e. PC - Phone - Switch? Actually it is a legacy PBX - Asterisk integration Legacy Handset -- Mitel SX 200 -- Asterisk -- Switch -- Router The calls come inbound over the internet as SIP to Asterisk and are routed into the Mitels ACD queue system where the user picks it up. Then you don't have a local LAN problem. You have a QoS issue with your WAN connection. Since I doubt your ISP has QoS on the link you'll get audio issues. Unless you have audio issues between calls that don't hit the router, in which case I have no idea what to suggest. -- Eric Wieling * BTEL Consulting * 504-210-3699 x2120 r: Generate a ringing tone for the calling party, passing no audio from the called channel(s) until one answers. Use with care and don't insert this by default into all your dial statements as you are killing call progress information for the user. Really, you almost certainly do not want to use this. Asterisk will generate ring tones automatically where it is appropriate to do so. r makes it go the next step and additionally generate ring tones where it is probably not appropriate to do so. ___ 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 General Question
Half duplex by itself doesn't hurt (depends in number of calls and etc really, but anyway...) What is a killer for VOIP is duplex mismatch. If you have autonegotiation enabled, and your peer (the switch ?) has autoneg off, and 100/Full-duplex hard coded, you WILL have a duplex mismatch. And this is as per the spec Geoff Manning wrote: Eric Wieling aka ManxPower wrote: In my experience, for local LAN audio issues, duplex problems are the problem, not LAN traffic. Rock on! I am in half duplex mode: serv01:~# ethtool eth0 Settings for eth0: Supported ports: [ MII ] Supported link modes: 10baseT/Half 10baseT/Full 100baseT/Half 100baseT/Full 1000baseT/Half 1000baseT/Full Supports auto-negotiation: Yes Advertised link modes: 10baseT/Half 10baseT/Full 100baseT/Half 100baseT/Full 1000baseT/Half 1000baseT/Full Advertised auto-negotiation: Yes Speed: 100Mb/s Duplex: Half Port: Twisted Pair PHYAD: 1 Transceiver: internal Auto-negotiation: on Supports Wake-on: g Wake-on: d Current message level: 0x00ff (255) Link detected: yes This could help solve a lot of quality issues. ___ 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 General Question
Julio Arruda wrote: Half duplex by itself doesn't hurt (depends in number of calls and etc really, but anyway...) What is a killer for VOIP is duplex mismatch. If you have autonegotiation enabled, and your peer (the switch ?) has autoneg off, and 100/Full-duplex hard coded, you WILL have a duplex mismatch. And this is as per the spec We'll have only about 5 or 6 concurrent ulaw/alaw calls. And the server is on the LAN with all the other workstations. Here is my output of ifconfig where you can see alot of collisions. eth0 Link encap:Ethernet HWaddr 00:13:20:17:DA:84 inet addr:172.16.64.15 Bcast:172.16.255.255 Mask:255.255.240.0 inet6 addr: fe80::213:20ff:fe17:da84/64 Scope:Link UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 RX packets:45521263 errors:0 dropped:0 overruns:0 frame:247 TX packets:46135708 errors:0 dropped:0 overruns:0 carrier:0 collisions:70538 txqueuelen:1000 RX bytes:1261961045 (1.1 GiB) TX bytes:1711703099 (1.5 GiB) Interrupt:177 Thanks, Geoff ___ 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 General Question
On 14:35, Tue 09 Aug 05, Geoff Manning wrote: Julio Arruda wrote: Half duplex by itself doesn't hurt (depends in number of calls and etc really, but anyway...) What is a killer for VOIP is duplex mismatch. If you have autonegotiation enabled, and your peer (the switch ?) has autoneg off, and 100/Full-duplex hard coded, you WILL have a duplex mismatch. And this is as per the spec We'll have only about 5 or 6 concurrent ulaw/alaw calls. And the server is on the LAN with all the other workstations. Here is my output of ifconfig where you can see alot of collisions. eth0 Link encap:Ethernet HWaddr 00:13:20:17:DA:84 inet addr:172.16.64.15 Bcast:172.16.255.255 Mask:255.255.240.0 inet6 addr: fe80::213:20ff:fe17:da84/64 Scope:Link UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 RX packets:45521263 errors:0 dropped:0 overruns:0 frame:247 TX packets:46135708 errors:0 dropped:0 overruns:0 carrier:0 collisions:70538 txqueuelen:1000 RX bytes:1261961045 (1.1 GiB) TX bytes:1711703099 (1.5 GiB) Interrupt:177 We had the same, till we replaced the switch with a new Cisco 2950. Now we have no collisions nor errors after 300 days of uptime. Check the cables, switch and NIC. Michiel ___ 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 General Question
How much of an impact can/does local network traffic have on call quality? Would opening large files on local servers affect call quality? We are running QoS on the router but that will only prioritize traffic in/out of the network. Sure it can. If you have a network segment that's fully saturated and you're also pushing VOIP data over that segment you'll have problems. In practice most networks are not that busy, but it can happen. If your phones, switch and NICs are VLAN capable you can setup a dedicated VLAN for the voice traffic and ensure that it gets priority. A vlan won't fix anything other then a minor step towards improving security. (And, it really is a minor step.) We do a lot of network performance assessments throughout the US, and I can't begin to count the number of corporations/institutions that don't have a clue how many packets are dropped by their layer-2 switches simply because they don't monitor the key snmp oid. The key is watching for discarded packets on outbound ports. (The majority of network managers believe their layer-2 switches have buffers just like layer-3 boxes, and the majority do not have buffers. The most simple example is two PC's attached to the same switch sending multiple packets at 100 meg, and the outbound (trunk) port running at 100 meg. The 200 meg of inbound data (to the switch) will frequently congest the outbound port causing the switch to drop (discard) packets. In real time, that can be as few as 5 or 10 packets from each PC, if they happen at the same time. (Note: many of the newer switches on the market today do have some amount of buffering, but the majority of the two to five year old switches do not.) For those that would really like to argue that point, take the covers off your switch, identify the chip set, and read the techie detail in the spec sheets. Or, do some simple tests by trying to overload an outbound port and see what happens. Essentially, if a switch supports QoS properly, it _will_ have some amount of buffering. QoS will help, but if the outbound load is to great, the traffic is still going to cause the switch to run out of buffer space and drop the packets. ___ 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] QoS windows client - cFosSpeed
I know this isn't directly related to * but I found it works very well in my voip environment. Check out cFosSpeed @ www.cfos.de. It gives you QoS based on applications and also seems to have increased my network throughput. Cheers, S. smime.p7s Description: S/MIME cryptographic signature ___ 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] QoS settings of the SIPURA ATA
Hi All, There are two option in QoS settings of the SIPURA ATA. ( I can't just remember them). please tell me what is better and which one should I choose for my DSL line (128kbps) with a small LAN. thank you 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] QoS settings of the SIPURA ATA
Hi All, There are two option in QoS settings of the SIPURA ATA. ( I can't just remember them). please tell me what is better and which one should I choose for my DSL line (128kbps) with a small LAN. thank you 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] QOS of VoIP
Hi All From wherewe can get the data for 1) ASR on various countries 2) Average Call drop on VoIP 3) Average Call Quality This we require to get an idea of what types of problem normally users use to face on voip and what is the average percentage of those problems. Pls. help me if anybody have the factsheet for various service provider on these paramaters Thanks Regards Ritesh Jalan ___ 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
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
[Asterisk-Users] Qos betwenn WIFI machines in LAN? oh323?
How do you do Qos between two machines when the bandwidth changes such as with WIFI? I normally get about 15 Mbit/s but this changes between 9 to 19 Mbits/s at times. Also, I use ohphone. How does one prioritize these oh323 packets or tag them for higher priority? I also have mythtv running in some machines, and this causes choppy voip when I have mythtv streaming at the same time from the same voip box. Thanks in advance. ___ 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 betwenn WIFI machines in LAN? oh323?
-Original Message- From: cmisip [mailto:[EMAIL PROTECTED] Sent: Friday, May 06, 2005 9:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Qos betwenn WIFI machines in LAN? oh323? How do you do Qos between two machines when the bandwidth changes such as with WIFI? I normally get about 15 Mbit/s but this changes between 9 to 19 Mbits/s at times. Also, I use ohphone. How does one prioritize these oh323 packets or tag them for higher priority? I also have mythtv running in some machines, and this causes choppy voip when I have mythtv streaming at the same time from the same voip box. Thanks in advance. You, and the others discusing QoS and MTU issues may find http://www.lartc.org (Linux Advanced Routing Traffic Control HOWTO) useful. Of particular note should be Chapter 9.2.2 (Simple, classless Queueing Disciplines: Token Bucket Filter) and Chapter 15.9 (Cookbook - The Ultimate Traffic Conditioner: Low Latency, Fast Up Downloads). Also http://www.tldp.org/HOWTO/ADSL-Bandwidth-Management-HOWTO/index.html (ADSL Bandwidth Management HOWTO) may provide useful knowledge. Enjoy. Kris Boutilier Information Services Coordinator Sunshine Coast Regional District ___ 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
[Asterisk-Users] QoS for improvements
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] QoS Help and survey
Hi - We've been using IAX forwards between sites for a little while now (with centralized VM). For the most part, it is fine, but I have some very minor, yet persistent QoS issues on calls over the IAX forwards. For most normal calls, there are very occasional minor glitches, just an infrequent popping sound. It is something most of my users don't really care about, although it is a minor annoyance for some of them. Strangely, the problem is significantly more noticeable on voicemail and directory calls, and it is not limited to just pops. I also get large drops and strange metallic sounding echos and repeated sounds (voicema-ma-ma-ma-ma-mail - a la Max Headroom, for those that remember that). The issue isn't horrible, but it is a little weird and annoying. It generally only happens when network traffic between the sites is heavy. So, my survey question is - Is this normal? Should I expect to be able to get PSTN quality calls over these IAX forwards, or are some audio glitches just part of the package? I use a commercial VoIP service at home, and I don't have any of these issues, so I'm guessing it must be something in my network or setup. Our setup: - CVS HEAD from about a month ago on all machines (problem was also there with CVS HEAD as far back as 11/04) - Late Model Dell Servers 1600SC and SC420's - Cisco Routers - 1751 and 1721's (using Low Latency Queueing, matched to UDP 4569) - T1's - 10/100 Switches - 10/100 hubs at one site (is this a problem for anyone?) - SIP phones (Polycom IP300, IP500 and IP600, Snom 190) Thanks, Noah ___ 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 Routers
I run m0n0wall (http://m0n0.ch/wall) on a Soekris 4501 embedded PC (http://www.soekris.com). Very tweakable. Under $200. Michael On Fri, 22 Apr 2005 10:42:20 -0700, Max Clark wrote: Hi all, I am looking for good (sub $200 dollars) routers to support VoIP installations. What is available at this point? I've used Netscreen and Checkpoint in the past, they are just too much overkill for this application. TIA, Max -- Max Clark max [at] clarksys.com http://www.clarksys.com ___ 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 Routers
On Apr 22, 2005, at 2:54 PM, Jay Milk wrote: Sveasoft is useless -- use hyperWRT instead. -Original Message- From: Gregory Wiktor - ADCom Corp. [mailto:[EMAIL PROTECTED] Sent: Friday, April 22, 2005 4:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] QOS Routers How about a linksys wrt54g with sveasoft firmware? Has some shaping and many other nice features... Jay: can you elaborate on your standpoint on the svea firmware? thx /rg ___ 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 Routers
You may want to check out edgewaternetworks. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Max Clark Sent: Friday, April 22, 2005 11:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] QOS Routers Hi all, I am looking for good (sub $200 dollars) routers to support VoIP installations. What is available at this point? I've used Netscreen and Checkpoint in the past, they are just too much overkill for this application. TIA, Max -- Max Clark max [at] clarksys.com http://www.clarksys.com ___ 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 Routers
How about a linksys wrt54g with sveasoft firmware? Has some shaping and many other nice features... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Max Clark Sent: Friday, April 22, 2005 1:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] QOS Routers Hi all, I am looking for good (sub $200 dollars) routers to support VoIP installations. What is available at this point? I've used Netscreen and Checkpoint in the past, they are just too much overkill for this application. TIA, Max -- Max Clark max [at] clarksys.com http://www.clarksys.com ___ 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 Routers
You may want to check out edgewaternetworks. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Max Clark Sent: Friday, April 22, 2005 11:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] QOS Routers Hi all, I am looking for good (sub $200 dollars) routers to support VoIP installations. What is available at this point? I've used Netscreen and Checkpoint in the past, they are just too much overkill for this application. TIA, Max -- Max Clark max [at] clarksys.com http://www.clarksys.com ___ 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 Routers
Max Clark wrote: Hi all, I am looking for good (sub $200 dollars) routers to support VoIP installations. What is available at this point? I've used Netscreen and Checkpoint in the past, they are just too much overkill for this application. I was using a boot off cd distro called Devil linux... with a hand written shaping and QOS script.. but I've since moved it over to a linksys wrt54gs box running the openwrt firmware. I manage the firewall side with the fwbuilder package from my windows laptop. It also supports pptp and openvpn tunnels... and I have a small instance of asterisk running... in addition to the standard stuff like dns cache and dyndns client. Jared ___ 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 Routers
Linksys with HyperWRT -- sub $100. You can make one of the LAN ports a QOS port and don't even have to worry about setting up protocols. -Original Message- From: Max Clark [mailto:[EMAIL PROTECTED] Sent: Friday, April 22, 2005 12:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] QOS Routers Hi all, I am looking for good (sub $200 dollars) routers to support VoIP installations. What is available at this point? I've used Netscreen and Checkpoint in the past, they are just too much overkill for this application. TIA, Max -- Max Clark max [at] clarksys.com http://www.clarksys.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/aster isk-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 Routers
Sveasoft is useless -- use hyperWRT instead. -Original Message- From: Gregory Wiktor - ADCom Corp. [mailto:[EMAIL PROTECTED] Sent: Friday, April 22, 2005 4:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] QOS Routers How about a linksys wrt54g with sveasoft firmware? Has some shaping and many other nice features... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Max Clark Sent: Friday, April 22, 2005 1:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] QOS Routers Hi all, I am looking for good (sub $200 dollars) routers to support VoIP installations. What is available at this point? I've used Netscreen and Checkpoint in the past, they are just too much overkill for this application. TIA, Max -- Max Clark max [at] clarksys.com http://www.clarksys.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/aster isk-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 ___ 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] QOS Routers
Hi all, I am looking for good (sub $200 dollars) routers to support VoIP installations. What is available at this point? I've used Netscreen and Checkpoint in the past, they are just too much overkill for this application. TIA, Max -- Max Clark max [at] clarksys.com http://www.clarksys.com ___ 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] qos test
Good day all I'm looking for a type of QOS test tool(software) I want to test if a link is good enough for voip and test witch ones will be the best..ens any ideas ___ 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] QoS TOS numbers and Cisco IOS
Does anyone know how setting the TOS bits in iax.conf corresponds to the Cisco TOS types? For example, if I set: tos=0x04 in iax.conf, and on the Cisco, I use: access-list 110 permit ip any any tos 4 I can't get the Cisco to match any packets. I've tried various combinations of numbers on both asterisk and the cisco. I've also tried hex to decimal conversion. I just can't get the Cisco to see the TOS bits that I set in iax.conf. Thanks, Noah ___ 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 TOS numbers and Cisco IOS
Does anyone know how setting the TOS bits in iax.conf corresponds to the Cisco TOS types? For example, if I set: tos=0x04 in iax.conf, and on the Cisco, I use: access-list 110 permit ip any any tos 4 I can't get the Cisco to match any packets. I've tried various combinations of numbers on both asterisk and the cisco. I've also tried hex to decimal conversion. I just can't get the Cisco to see the TOS bits that I set in iax.conf. Here's what I'm using. sip.conf: tos=0x18 ;lowdelay ;sets ip tos bits (=lowdelay, throughput) iax.conf: tos=lowdelay Cisco: class-map match-all voice-rtp match access-group 103 access-list 103 permit ip any any tos min-delay access-list 103 permit ip any any tos 12 C1750#show access-list 103 Extended IP access list 103 permit ip any any tos min-delay (2077271 matches) permit ip any any tos 12 (651833 matches) The NAI Sniffer does a better job of showing the bits. Here's two samples for the above: sip packet (tos=0x18): IP: Type of service = 18 IP: 000. = routine IP: ...1 = low delay IP: 1... = high throughput IP: .0.. = normal reliability IP: ..0. = ECT bit - transport protocol will ignore the CE bit IP: ...0 = CE bit - no congestion iax packet (tos=lowdelay): IP: Type of service = 10 IP: 000. = routine IP: ...1 = low delay IP: 0... = normal throughput IP: .0.. = normal reliability IP: ..0. = ECT bit - transport protocol will ignore the CE bit IP: ...0 = CE bit - no congestion Study the above and the bits become very clear. :) ___ 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 TOS numbers and Cisco IOS
tos=0xb8 will set the the packet to be DSCP EF (Cisco likes to use DSCP) Rich Adamson wrote: Does anyone know how setting the TOS bits in iax.conf corresponds to the Cisco TOS types? For example, if I set: tos=0x04 in iax.conf, and on the Cisco, I use: access-list 110 permit ip any any tos 4 I can't get the Cisco to match any packets. I've tried various combinations of numbers on both asterisk and the cisco. I've also tried hex to decimal conversion. I just can't get the Cisco to see the TOS bits that I set in iax.conf. Here's what I'm using. sip.conf: tos=0x18 ;lowdelay ;sets ip tos bits (=lowdelay, throughput) iax.conf: tos=lowdelay Cisco: class-map match-all voice-rtp match access-group 103 access-list 103 permit ip any any tos min-delay access-list 103 permit ip any any tos 12 C1750#show access-list 103 Extended IP access list 103 permit ip any any tos min-delay (2077271 matches) permit ip any any tos 12 (651833 matches) The NAI Sniffer does a better job of showing the bits. Here's two samples for the above: sip packet (tos=0x18): IP: Type of service = 18 IP: 000. = routine IP: ...1 = low delay IP: 1... = high throughput IP: .0.. = normal reliability IP: ..0. = ECT bit - transport protocol will ignore the CE bit IP: ...0 = CE bit - no congestion iax packet (tos=lowdelay): IP: Type of service = 10 IP: 000. = routine IP: ...1 = low delay IP: 0... = normal throughput IP: .0.. = normal reliability IP: ..0. = ECT bit - transport protocol will ignore the CE bit IP: ...0 = CE bit - no congestion Study the above and the bits become very clear. :) ___ 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 -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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
