[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 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
[Asterisk-Users] QoS for Asterisk with ISA 2004 Server?
I'm thinking of using ISA 2004 for QoS with asterisk. Right now I'm using FreeBSD with Packet Filter as a firewall and traffic shaper to do QoS for my Asterisk setup. I wanted to explore other options, like ISA 2004. Has anyone implemented it for asterisk? Any gotchas? ___ 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 / Cisco / Asterisk
Sorry for the late, late reply, but I don't follow the -users list closely. On Tue, 2005-01-04 at 10:43 -0600, [EMAIL PROTECTED] wrote: What's wrong with doing it by port? If it is possible that something else out there may use the same TOS flags as Asterisk, by prioritizing port 4569 (IAX2 protocol) you know for sure that the only packets in that queue are VoIP traffic. Also, what about your incoming traffic? Are the TOS flags correct there? I'm not saying that TOS is bad, just that as you've seen, it can get changed along the way. I'm using port number to separate traffic and it is working great. Well, in a sense, we are both correct. You are looking at the problem from the perspective of an edge router. At the edge of your network, you can't trust the incoming QOS markings, so you need to use an ACL of some sort to differentiate priority traffic from non-priority traffic. However, inside the network, when you can (mostly) trust that packets have been generated with the correct QOS markings by the orginating device, internal routers/switches can use the QOS marking (be it the TOS, DiffServ markings, 802.1p priorities, etc.) to prioritize traffic. I'd be willing to bet that switches (and maybe even some routers) can prioritize based upon QOS markings more efficiently that they can run packets through ACLs. This is especially needed where traffic volumes are large. So, inside your network you need to examine the configuration of pretty much every device to make sure that they don't mess with the QOS markings where they aren't supposed to. Jeff signature.asc Description: This is a digitally signed message part ___ 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 / Cisco / Asterisk
Title: Message Yes yes, we've been through all that actually :-) We did find out it was one of the 3550's reseting the TOS. -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 04, 2005 2:40 PMTo: asterisk-users@lists.digium.comSubject: RE: [Asterisk-Users] QOS / Cisco / Asterisksnip What's wrong with doing it by port? We're actually using SIP to terminate calls, going by rtp.conf the portscould range several thousand ports. What we're going for is onlyhonoring TOS for that particular customer, luckily these are T1customers hosted on our routers. They understand that their firewallscannot pass TOS, if they do (ie: we packet sniff and see this) thenthey're on their own.In a nutshell we wanted to avoid using hardcoded ports, what if say agame server was in that port range (and used udp lol), you would berather screwed. /snip Ahh OK. Well, how about configuring a laptop with ethereal (http://www.ethereal.com/) and capturing the packets you have in mind? It even runs on Windows. :p It's pretty easy to specify a particular destination or so, for limiting which traffic you sniff. You could use an old hub and start plugging the laptop in between routers using the hub so it can capture the packets. Should be fairly quick to isolate which router is modifying the TOS value. Just an idea... of course you have to have physical access to the network... HTH, -Ron ___ 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 / Cisco / Asterisk
Yes yes, your right. I forget these switches are smart!!! ;-) -Original Message- From: Julio Arruda [mailto:[EMAIL PROTECTED] Sent: Monday, January 03, 2005 4:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] QOS / Cisco / Asterisk Matt Schulte wrote: We're trying to PQ (Priority Queue) packets on a Cisco using ACL's. What we're trying to avoid is hardcoding the IP address in the ACL. We were trying to match by TOS set by Asterisk however it seems we've run into a snag where the packet TOS tends to get reset somewhere on our network. Has anyone had this issue? We're running Cisco everywhere inbetween (even the switches). Is there an alternative way to match these? We've thought of by port but that's kind of ad-hoc IMHO. I know some LAN switching devices, in a default QoS configuration, would treat ports as diffserv untrusted ports, or access ports, meaning, the DSCP (a reuse of the TOS also) in packets inbound at that port are not to be trusted. Have you looked at your switches documentation ? Asterisk1 -- 3560 -- 2600 -- (T1) -- 7500 -- 2900 -- 3550 -- Asterisk2 Sniff: (note the dumps between the 2 machines are diff times however they show the same occurance) Asterisk1: 1.1.1.1 09:09:10.019191 IP (tos 0x10, ttl 64, id 58, offset 0, flags [DF], proto 17, length: 60) 1.1.1.1.12056 1.1.1.2.19726: [no cksum] UDP, length 32 09:09:10.030146 IP (tos 0x0, ttl 62, id 63, offset 0, flags [DF], proto 17, length: 60) 1.1.1.2.19726 1.1.1.1.12056: [no cksum] UDP, length 32 Asterisk2: Dump on 206.80.70.55 09:34:34.418386 IP (tos 0x0, ttl 62, id 261, offset 0, flags [DF], proto 17, length: 60) 1.1.1.1.14796 1.1.1.2.18996: [no cksum] UDP, length 32 09:34:34.422974 IP (tos 0x10, ttl 64, id 273, offset 0, flags [DF], proto 17, length: 60) 1.1.1.2.18996 1.1.1.1.14796: [no cksum] UDP, length 32 ___ 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 / Cisco / Asterisk
On Mon, 2005-01-03 at 13:53 -0600, Matt Schulte wrote: We're trying to PQ (Priority Queue) packets on a Cisco using ACL's. What we're trying to avoid is hardcoding the IP address in the ACL. We were trying to match by TOS set by Asterisk however it seems we've run into a snag where the packet TOS tends to get reset somewhere on our network. Has anyone had this issue? We're running Cisco everywhere inbetween (even the switches). Is there an alternative way to match these? We've thought of by port but that's kind of ad-hoc IMHO. If the TOS is getting reset somewhere out there you need to go through all of your switches and make sure that none of them are messing with the TOS. Unfortunately doing QOS on Cisco switches is a black art as the necessary commands depend on the hardware and the IOS version (or CatOS version if you are unlucky). Check the documentation for your switches for the mls qos trust command. Cisco routers, on the other hand, don't mess with IP TOS/DSCP labels unless you specifically ask them to. Jeff signature.asc Description: This is a digitally signed message part ___ 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 / Cisco / Asterisk
snip We're trying to PQ (Priority Queue) packets on a Cisco using ACL's. What we're trying to avoid is hardcoding the IP address in the ACL. We were trying to match by TOS set by Asterisk however it seems we've run into a snag where the packet TOS tends to get reset somewhere on our network. Has anyone had this issue? We're running Cisco everywhere inbetween (even the switches). Is there an alternative way to match these? We've thought of by port but that's kind of ad-hoc IMHO. If the TOS is getting reset somewhere out there you need to go through all of your switches and make sure that none of them are messing with the TOS. Unfortunately doing QOS on Cisco switches is a black art as the necessary commands depend on the hardware and the IOS version (or CatOS version if you are unlucky). Check the documentation for your switches for the mls qos trust command. Cisco routers, on the other hand, don't mess with IP TOS/DSCP labels unless you specifically ask them to. /snip What's wrong with doing it by port? If it is possible that something else out there may use the same TOS flags as Asterisk, by prioritizing port 4569 (IAX2 protocol) you know for sure that the only packets in that queue are VoIP traffic. Also, what about your incoming traffic? Are the TOS flags correct there? I'm not saying that TOS is bad, just that as you've seen, it can get changed along the way. I'm using port number to separate traffic and it is working great. -Ron___ 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 / Cisco / Asterisk
What's wrong with doing it by port? We're actually using SIP to terminate calls, going by rtp.conf the ports could range several thousand ports. What we're going for is only honoring TOS for that particular customer, luckily these are T1 customers hosted on our routers. They understand that their firewalls cannot pass TOS, if they do (ie: we packet sniff and see this) then they're on their own. In a nutshell we wanted to avoid using hardcoded ports, what if say a game server was in that port range (and used udp lol), you would be rather screwed. same TOS flags as Asterisk, by prioritizing port 4569 (IAX2 protocol) you know for sure that the only packets in that queue are VoIP traffic. Also, what about your incoming traffic? Are the TOS flags correct there? I'm not saying that TOS is bad, just that as you've seen, it can get changed along the way. I'm using port number to separate traffic and it is working great. -Ron ___ 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 / Cisco / Asterisk
We're trying to PQ (Priority Queue) packets on a Cisco using ACL's. --- You do not want to use PQ for voice QOS. You will still receive far too much jitter. Instead configure LLQ which was specifically designed for voice scheduling on an interface. Aside from being designed for voice, LLQ also allows you to create lower priority queues for other traffic without running into queue starvation problems. For a complete description on designing and configuring Cisco networks for voice QOS see: http://www.cisco.com/application/pdf/en/us/guest/netsol/ns17/c649/ccmigration_09186a00800d67ed.pdf ___ 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 / Cisco / Asterisk
snip What's wrong with doing it by port? We're actually using SIP to terminate calls, going by rtp.conf the ports could range several thousand ports. What we're going for is only honoring TOS for that particular customer, luckily these are T1 customers hosted on our routers. They understand that their firewalls cannot pass TOS, if they do (ie: we packet sniff and see this) then they're on their own. In a nutshell we wanted to avoid using hardcoded ports, what if say a game server was in that port range (and used udp lol), you would be rather screwed. /snip Ahh OK. Well, how about configuring a laptop with ethereal (http://www.ethereal.com/) and capturing the packets you have in mind? It even runs on Windows. :p It's pretty easy to specify a particular destination or so, for limiting which traffic you sniff. You could use an old hub and start plugging the laptop in between routers using the hub so it can capture the packets. Should be fairly quick to isolate which router is modifying the TOS value. Just an idea... of course you have to have physical access to the network... HTH, -Ron ___ 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 / Cisco / Asterisk
We're trying to PQ (Priority Queue) packets on a Cisco using ACL's. What we're trying to avoid is hardcoding the IP address in the ACL. We were trying to match by TOS set by Asterisk however it seems we've run into a snag where the packet TOS tends to get reset somewhere on our network. Has anyone had this issue? We're running Cisco everywhere inbetween (even the switches). Is there an alternative way to match these? We've thought of by port but that's kind of ad-hoc IMHO. Asterisk1 -- 3560 -- 2600 -- (T1) -- 7500 -- 2900 -- 3550 -- Asterisk2 Sniff: (note the dumps between the 2 machines are diff times however they show the same occurance) Asterisk1: 1.1.1.1 09:09:10.019191 IP (tos 0x10, ttl 64, id 58, offset 0, flags [DF], proto 17, length: 60) 1.1.1.1.12056 1.1.1.2.19726: [no cksum] UDP, length 32 09:09:10.030146 IP (tos 0x0, ttl 62, id 63, offset 0, flags [DF], proto 17, length: 60) 1.1.1.2.19726 1.1.1.1.12056: [no cksum] UDP, length 32 Asterisk2: Dump on 206.80.70.55 09:34:34.418386 IP (tos 0x0, ttl 62, id 261, offset 0, flags [DF], proto 17, length: 60) 1.1.1.1.14796 1.1.1.2.18996: [no cksum] UDP, length 32 09:34:34.422974 IP (tos 0x10, ttl 64, id 273, offset 0, flags [DF], proto 17, length: 60) 1.1.1.2.18996 1.1.1.1.14796: [no cksum] UDP, length 32 ___ 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 / Cisco / Asterisk
Matt Schulte wrote: We're trying to PQ (Priority Queue) packets on a Cisco using ACL's. What we're trying to avoid is hardcoding the IP address in the ACL. We were trying to match by TOS set by Asterisk however it seems we've run into a snag where the packet TOS tends to get reset somewhere on our network. Has anyone had this issue? We're running Cisco everywhere inbetween (even the switches). Is there an alternative way to match these? We've thought of by port but that's kind of ad-hoc IMHO. I know some LAN switching devices, in a default QoS configuration, would treat ports as diffserv untrusted ports, or access ports, meaning, the DSCP (a reuse of the TOS also) in packets inbound at that port are not to be trusted. Have you looked at your switches documentation ? Asterisk1 -- 3560 -- 2600 -- (T1) -- 7500 -- 2900 -- 3550 -- Asterisk2 Sniff: (note the dumps between the 2 machines are diff times however they show the same occurance) Asterisk1: 1.1.1.1 09:09:10.019191 IP (tos 0x10, ttl 64, id 58, offset 0, flags [DF], proto 17, length: 60) 1.1.1.1.12056 1.1.1.2.19726: [no cksum] UDP, length 32 09:09:10.030146 IP (tos 0x0, ttl 62, id 63, offset 0, flags [DF], proto 17, length: 60) 1.1.1.2.19726 1.1.1.1.12056: [no cksum] UDP, length 32 Asterisk2: Dump on 206.80.70.55 09:34:34.418386 IP (tos 0x0, ttl 62, id 261, offset 0, flags [DF], proto 17, length: 60) 1.1.1.1.14796 1.1.1.2.18996: [no cksum] UDP, length 32 09:34:34.422974 IP (tos 0x10, ttl 64, id 273, offset 0, flags [DF], proto 17, length: 60) 1.1.1.2.18996 1.1.1.1.14796: [no cksum] UDP, length 32 ___ 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 in asterisk
On 11 Jul 2004 at 19:16, Rich Adamson wrote: QoS is most certainly an issue when making the decision to move off the PSTN. Is the performance of your VoIP system going to be comparable to the performance of your PSTN system? Sounds like a reasonable question to me. Not trying to get in the middle of whatever argument you're trying to make, the poster's original question (although probably not worded all that clear) can be answered by... no, asterisk cannot make a decision to route calls via a second path due to quality issues on some first choice path. Well...you could run an agi to check ping time for 1 sec and then if the differences are too much or the overall amount is too high, then use the POTS line... Matt Riddell ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 in asterisk
how do you ping a TDM connection ? On Mon, 2004-07-12 at 11:43, [EMAIL PROTECTED] wrote: On 11 Jul 2004 at 19:16, Rich Adamson wrote: QoS is most certainly an issue when making the decision to move off the PSTN. Is the performance of your VoIP system going to be comparable to the performance of your PSTN system? Sounds like a reasonable question to me. Not trying to get in the middle of whatever argument you're trying to make, the poster's original question (although probably not worded all that clear) can be answered by... no, asterisk cannot make a decision to route calls via a second path due to quality issues on some first choice path. Well...you could run an agi to check ping time for 1 sec and then if the differences are too much or the overall amount is too high, then use the POTS line... Matt Riddell ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 in asterisk
On Monday 12 July 2004 05:43, [EMAIL PROTECTED] wrote: Not trying to get in the middle of whatever argument you're trying to make, the poster's original question (although probably not worded all that clear) can be answered by... no, asterisk cannot make a decision to route calls via a second path due to quality issues on some first choice path. Well...you could run an agi to check ping time for 1 sec and then if the differences are too much or the overall amount is too high, then use the POTS line... Why not just work with qualify? If the connection is too lagged * won't make the call through it (although if the link BECOMES laggy it will continue to use the connection). -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 in asterisk
On 12 Jul 2004 at 14:06, Michael Bielicki wrote: how do you ping a TDM connection ? Sorry, where does it say this is regarding a TDM connection? I use IAX trunking and a ping script to check times and fluctuations to my remote offices. Matt Riddell On Mon, 2004-07-12 at 11:43, [EMAIL PROTECTED] wrote: On 11 Jul 2004 at 19:16, Rich Adamson wrote: QoS is most certainly an issue when making the decision to move off the PSTN. Is the performance of your VoIP system going to be comparable to the performance of your PSTN system? Sounds like a reasonable question to me. Not trying to get in the middle of whatever argument you're trying to make, the poster's original question (although probably not worded all that clear) can be answered by... no, asterisk cannot make a decision to route calls via a second path due to quality issues on some first choice path. Well...you could run an agi to check ping time for 1 sec and then if the differences are too much or the overall amount is too high, then use the POTS line... ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 in asterisk
On 12 Jul 2004 at 8:22, Andrew Kohlsmith wrote: On Monday 12 July 2004 05:43, [EMAIL PROTECTED] wrote: Not trying to get in the middle of whatever argument you're trying to make, the poster's original question (although probably not worded all that clear) can be answered by... no, asterisk cannot make a decision to route calls via a second path due to quality issues on some first choice path. Well...you could run an agi to check ping time for 1 sec and then if the differences are too much or the overall amount is too high, then use the POTS line... Why not just work with qualify? If the connection is too lagged * won't make the call through it (although if the link BECOMES laggy it will continue to use the connection). Qualify will only stop the call going through if for example the ping is above 200ms. I find most of my problems come from fluctuating ping times (~100ms) than from a stable high ping. Matt Riddell ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 in asterisk
Doesn't make any difference 'how' one might ping a remote site, ping will never qualify the Quality of the channel between two points. It will only suggest its up/down and possibly the delay at that specific point in time. Has nothing to do with whether packets were dropped or delayed some milliseconds before or after the ping, and the ping pkt would never be subjected to any positive QoS parameters implemented in the point-to-point network infrastructure. A large number of ISP's block icmp pkts anyway (for other reasons), so its not a reasonable way to determine anything. how do you ping a TDM connection ? On Mon, 2004-07-12 at 11:43, [EMAIL PROTECTED] wrote: On 11 Jul 2004 at 19:16, Rich Adamson wrote: QoS is most certainly an issue when making the decision to move off the PSTN. Is the performance of your VoIP system going to be comparable to the performance of your PSTN system? Sounds like a reasonable question to me. Not trying to get in the middle of whatever argument you're trying to make, the poster's original question (although probably not worded all that clear) can be answered by... no, asterisk cannot make a decision to route calls via a second path due to quality issues on some first choice path. Well...you could run an agi to check ping time for 1 sec and then if the differences are too much or the overall amount is too high, then use the POTS line... ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 in asterisk
Would you consider posting this this to the wiki? :) I think that would be great. On Mon, 2004-07-12 at 08:35, [EMAIL PROTECTED] wrote: On 12 Jul 2004 at 14:06, Michael Bielicki wrote: how do you ping a TDM connection ? Sorry, where does it say this is regarding a TDM connection? I use IAX trunking and a ping script to check times and fluctuations to my remote offices. Matt Riddell On Mon, 2004-07-12 at 11:43, [EMAIL PROTECTED] wrote: On 11 Jul 2004 at 19:16, Rich Adamson wrote: QoS is most certainly an issue when making the decision to move off the PSTN. Is the performance of your VoIP system going to be comparable to the performance of your PSTN system? Sounds like a reasonable question to me. Not trying to get in the middle of whatever argument you're trying to make, the poster's original question (although probably not worded all that clear) can be answered by... no, asterisk cannot make a decision to route calls via a second path due to quality issues on some first choice path. Well...you could run an agi to check ping time for 1 sec and then if the differences are too much or the overall amount is too high, then use the POTS line... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- respectfully, Joseph - (606) 477-2355 x140 --= ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 in asterisk
[EMAIL PROTECTED] wrote: I use IAX trunking and a ping script to check times and fluctuations to my remote offices. Could you share this AGI? - seems like a useful example :) Thanks a lot, F ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 in asterisk
On Tue, 13 Jul 2004 [EMAIL PROTECTED] wrote: Qualify will only stop the call going through if for example the ping is above 200ms. I find most of my problems come from fluctuating ping times (~100ms) than from a stable high ping. I agree that the overall delay isn't really the problem - jitter and packet loss are what causes the trouble. There really isn't currently anything in Asterisk which measures this - especially not when there is no active call using the path. The IAX2 jitter buffer code does know the amount of jitter - and could probably make this measurement available in a variable or something. And I propose to add similar jitter buffer code for SIP and other RTP-using protocols too. But I'm not really sure how the measurement can then be used effectively for call routing. I'd be interested in your ideas. Note that I observe that in my environment jitter and packet loss come and go over a timescale of seconds - this a result of sharing a narrowish pipe with a bunch of other traffic without any shaping to help the VOIP traffic. For this environment the real fix is to improve the network rather than do anything too complicated with *. (Not to say that *s jitter handling and packet-loss-concealment can't be improved - I've been working on that and I'm still busy). I'm about to ask for some help in gathering jitter stats from a bunch of users - perhaps you'd like to help with that. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 in asterisk
[EMAIL PROTECTED] wrote: On 12 Jul 2004 at 8:22, Andrew Kohlsmith wrote: Why not just work with qualify? If the connection is too lagged * won't make the call through it (although if the link BECOMES laggy it will continue to use the connection). Qualify will only stop the call going through if for example the ping is above 200ms. I find most of my problems come from fluctuating ping times (~100ms) than from a stable high ping. Well, the default for qualify = yes is 2000ms. You can specify any value you like, such as qualify = 1000. I find that the default is good enough for me; On the links I use, the ping time is either good, or it's very bad (usually because the remote service is down). Dial() will fall through if a link is down, so I use this, along with qualify, to specify a number of routes for each dialplan entry. This allows a transparent failover if a link is down. It doesn't help if a link goes down, or becomes laggy, when a call is in progress. Luckily that sort of thing is rare with the services I use. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 in asterisk
Does asterisk provide quality of service(QoS)? If it does, how do I use it? The reason why I ask is that I need to switch to use POTS should the internet connection becomes poor? Thanks, Jim ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 in asterisk
Does asterisk provide quality of service(QoS)? If it does, how do I use it? The reason why I ask is that I need to switch to use POTS should the internet connection becomes poor? Asterisk 'participates' in the qos process by allowing you to set TOS bits in the IP header. For example: In sip.conf tos=0x18;sets ip tos bits lowdelay throughput In iax.conf tos=lowdelay However, its up to your infrastructure equipment (eg, routers switches) to prioritize packets received, and send those on to the next hop following the objectives that you program into those devices. Not all devices support qos however. When using Internet connections, you can control qos for packets leaving your site, however you normally can't control how your ISP (and their providers) handle it (if at all) for incoming packets. Some ISPs are actually doing some qos, but most are not. There is no logic built into asterisk that would support a decision making process of chosing one path verses another based on past quality, etc. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 in asterisk
This is a very complex question. First, you have to ask about VoIP and QoS. This is because * uses VoIP protocols like UDP and RTP. In general, the QoS of VoIP is not as high as with the PSTN. Even so, call quality can be generally very good. Second, * does support features that support QoS, for example the IAX jitterbuffer setting. Paul Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training Does asterisk provide quality of service(QoS)? If it does, how do I use it? The reason why I ask is that I need to switch to use POTS should the internet connection becomes poor? Thanks, Jim ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 in asterisk
Both the question and the answer are not talking about QoS. From the Q, qos does not provide a measure of quality, it provides a system to allow you to request your data be handled according to priorities. From the A, qos is confused with the pstn.. qos is a feature of IP, that has nothing to do with the pstn. jitterbuffer isnt qos either, altho its important you get it right to provide good quality calls. qos is the tos options you can specify in the conf files but you need to combine that with routers from server to client that will honor the tos you set. deciding to switch from voip to pstn wouldnt be covered by qos, you would need to find other ways.. playing with options like 'qualify' to timeout poorly connected devices quickly is more like what you are trying to achieve. Steve On Sun, 11 Jul 2004, Paul Mahler wrote: This is a very complex question. First, you have to ask about VoIP and QoS. This is because * uses VoIP protocols like UDP and RTP. In general, the QoS of VoIP is not as high as with the PSTN. Even so, call quality can be generally very good. Second, * does support features that support QoS, for example the IAX jitterbuffer setting. Paul Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training Does asterisk provide quality of service(QoS)? If it does, how do I use it? The reason why I ask is that I need to switch to use POTS should the internet connection becomes poor? Thanks, Jim ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 in asterisk
At Astricon, I plan to cover QoS on FreeBSD using the pf firewall's class based queuing. This includes implementing classes to prioritize each of RTP, IAX, SIP, FTP, and others. Within each class packets can be prioritized based on whether TOS is set. I'm wondering whether this should be a tutorial rather than a talk. Anyone have presences? Cheers, Rich -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Sunday, July 11, 2004 3:06 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] QoS in asterisk Does asterisk provide quality of service(QoS)? If it does, how do I use it? The reason why I ask is that I need to switch to use POTS should the internet connection becomes poor? Asterisk 'participates' in the qos process by allowing you to set TOS bits in the IP header. For example: In sip.conf tos=0x18;sets ip tos bits lowdelay throughput In iax.conf tos=lowdelay However, its up to your infrastructure equipment (eg, routers switches) to prioritize packets received, and send those on to the next hop following the objectives that you program into those devices. Not all devices support qos however. When using Internet connections, you can control qos for packets leaving your site, however you normally can't control how your ISP (and their providers) handle it (if at all) for incoming packets. Some ISPs are actually doing some qos, but most are not. There is no logic built into asterisk that would support a decision making process of chosing one path verses another based on past quality, etc. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 in asterisk
Well, the question may not have been about QoS, but my answer certainly was. QoS is defined as The performance specification of a communications channel or system. (188) Note: QOS may be quantitatively indicated by channel or system performance parameters, such as signal-to-noise ratio (S/N), bit error ratio (BER), message throughput rate, and call blocking probability. Quality of Service (QoS) is a general term for an abstraction covering aspects of the non-functional behavior of a system, for example delay. I think what we have here is what we are going to see a lot of--cultures in collision. The PSTN folks had QoS issues long before it became an IP issue. I think what you are alluding to is routing specific IP QoS. IP supports QoS in the IP header, those pesky tos bits you were talking about. Asynchronous transfer mode (ATM) natively provides QoS. The IEEE 802.1p standard covers QoS in all IEEE 802-type networks. Even in networking, QoS is first and formost an abstraction before it becomes a specification. QoS is the ability of a network element (e.g. an application, a host or a router) to provide some level of assurance for consistent network data delivery. QoS is most certainly an issue when making the decision to move off the PSTN. Is the performance of your VoIP system going to be comparable to the performance of your PSTN system? Sounds like a reasonble question to me. Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen J. Wilcox Sent: Sunday, July 11, 2004 3:33 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] QoS in asterisk Both the question and the answer are not talking about QoS. From the Q, qos does not provide a measure of quality, it provides a system to allow you to request your data be handled according to priorities. From the A, qos is confused with the pstn.. qos is a feature of IP, that has nothing to do with the pstn. jitterbuffer isnt qos either, altho its important you get it right to provide good quality calls. qos is the tos options you can specify in the conf files but you need to combine that with routers from server to client that will honor the tos you set. deciding to switch from voip to pstn wouldnt be covered by qos, you would need to find other ways.. playing with options like 'qualify' to timeout poorly connected devices quickly is more like what you are trying to achieve. Steve On Sun, 11 Jul 2004, Paul Mahler wrote: This is a very complex question. First, you have to ask about VoIP and QoS. This is because * uses VoIP protocols like UDP and RTP. In general, the QoS of VoIP is not as high as with the PSTN. Even so, call quality can be generally very good. Second, * does support features that support QoS, for example the IAX jitterbuffer setting. Paul Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training Does asterisk provide quality of service(QoS)? If it does, how do I use it? The reason why I ask is that I need to switch to use POTS should the internet connection becomes poor? Thanks, Jim ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 in asterisk
QoS is most certainly an issue when making the decision to move off the PSTN. Is the performance of your VoIP system going to be comparable to the performance of your PSTN system? Sounds like a reasonble question to me. Not trying to get in the middle of whatever argument you're trying to make, the poster's original question (although probably not worded all that clear) can be answered by... no, asterisk cannot make a decision to route calls via a second path due to quality issues on some first choice path. The poster's question The reason why I ask is that I need to switch to use POTS should the internet connection become poor? is rather clear as to what his intent was. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 Asterisk
Kim C. Callis wrote: I was thinking of adding QoS to my Linux based router. I thought I would add all my IP phones and my * box into a VLAN, and then would do a QoS setup for that particular VLAN. Has anyone did any QoS setups for better performance? Has it made any change to the performance? Kim C. Callis http://www.lartc.org/ - complete howto http://www.docum.org/ - Very usefull examples ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] QoS for Asterisk
Jeremy McNamara wrote: No need to get all crazy with VLANs... just setup a Hierarchal Token Bucket Queue. That is if your edge device is linux Don't reckon you'd have any sample scripts in hand to help the intrepid but inexperienced HTB person make that happen? Thx. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] QoS for Asterisk
I was thinking of adding QoS to my Linux based router. I thought I would add all my IP phones and my * box into a VLAN, and then would do a QoS setup for that particular VLAN. Has anyone did any QoS setups for better performance? Has it made any change to the performance? Kim C. Callis
Re: [Asterisk-Users] QoS for Asterisk
No need to get all crazy with VLANs... just setup a Hierarchal Token Bucket Queue. That is if your edge device is linux Jeremy McNamara Kim C. Callis wrote: I was thinking of adding QoS to my Linux based router. I thought I would add all my IP phones and my * box into a VLAN, and then would do a QoS setup for that particular VLAN. Has anyone did any QoS setups for better performance? Has it made any change to the performance? Kim C. Callis ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
