[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
