VoIP quality issues
At a customer site we have a VoIP server running. The customer complains that the some voice conversations are not going well. Even with collegues next door.
The network guys have checked the network and told that there is no problem on the network. They tell that the voice server has not enough troughput or that the network card has an issue.
is there a software that i can install on the server to monitor the lan? i want to know what the max network troughput asked is. Not a measure like speedtest.net i want to know how much is asked when calling.
The server is windows based to i need a windows software.
The network guys have checked the network and told that there is no problem on the network. They tell that the voice server has not enough troughput or that the network card has an issue.
is there a software that i can install on the server to monitor the lan? i want to know what the max network troughput asked is. Not a measure like speedtest.net i want to know how much is asked when calling.
The server is windows based to i need a windows software.
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1) At a customer site we have a VoIP server running. The customer complains that the some voice conversations are not going well. Even with colleagues next door.
Proximity (closeness) has little to do with quality.
Depends on VOIP system being used + ISP providers of all participants.
2) The network guys have checked the network and told that there is no problem on the network.
To little information to guess.
You'd have to define all steps related to "checked the network" + all steps related to determining "there is no problem on the network".
3) They tell that the voice server has not enough throughput or that the network card has an issue.
See #2, same applies.
4) Is there a software that i can install on the server to monitor the lan?
No point in doing this till better information is provided.
5) I want to know what the max network throughput asked is. Not a measure like https://speedtest.net i want to know how much is asked when calling.
Using https://speedtest.net is your best step, because even if 2x people are sitting next to them, many VOIP systems will pass packets, back through the VOIP site, so voice quality will depend on speed of every network between all participants on a call.
If https://speedtest.net shows you have a slow connection, then unlikely you'll ever have good quality VOIP calls, especially when other participants on a call also have slow connections.
6) The server is windows based to i need a windows software.
First test is always to try a Skype call between 2x people.
If Skype works well, then the problem likely relates to the VOIP system being used.
If Skype works poorly, then problem likely relates to network speed of one or more call participants.
If Skype does work well, then using Skype for Business might be the answer.
Proximity (closeness) has little to do with quality.
Depends on VOIP system being used + ISP providers of all participants.
2) The network guys have checked the network and told that there is no problem on the network.
To little information to guess.
You'd have to define all steps related to "checked the network" + all steps related to determining "there is no problem on the network".
3) They tell that the voice server has not enough throughput or that the network card has an issue.
See #2, same applies.
4) Is there a software that i can install on the server to monitor the lan?
No point in doing this till better information is provided.
5) I want to know what the max network throughput asked is. Not a measure like https://speedtest.net i want to know how much is asked when calling.
Using https://speedtest.net is your best step, because even if 2x people are sitting next to them, many VOIP systems will pass packets, back through the VOIP site, so voice quality will depend on speed of every network between all participants on a call.
If https://speedtest.net shows you have a slow connection, then unlikely you'll ever have good quality VOIP calls, especially when other participants on a call also have slow connections.
6) The server is windows based to i need a windows software.
First test is always to try a Skype call between 2x people.
If Skype works well, then the problem likely relates to the VOIP system being used.
If Skype works poorly, then problem likely relates to network speed of one or more call participants.
If Skype does work well, then using Skype for Business might be the answer.
David is correct. We need much more information to make a guess at what is going on.
What is the cabling in the network? Is it recent Cat5e or Cat6, or early Cat3?
What switches are in use on the premises? Are they smart, or dumb? Who configured them?
What speed is the network supposedly running at?
What equipment is on premises as far as routing? Who configured it?
Please give us an overall view of what is on the premises. All equipment, all cabling, models in use, who the ISP is serving the premises.
What is the cabling in the network? Is it recent Cat5e or Cat6, or early Cat3?
What switches are in use on the premises? Are they smart, or dumb? Who configured them?
What speed is the network supposedly running at?
What equipment is on premises as far as routing? Who configured it?
Please give us an overall view of what is on the premises. All equipment, all cabling, models in use, who the ISP is serving the premises.
To add to Dr. Klahn's list will be WiFi hotspots + repeaters.
Also to add to my set of tests, if you have problems with Skype, best to plug 2x machines into the incoming router's first switch via Ethernet cables + retest Skype. This rules out any WiFi related problems.
So if you're using physical cabling, you must verify the cabling. If you're using any WiFi tech, you must verify your WiFi tech.
Also to add to my set of tests, if you have problems with Skype, best to plug 2x machines into the incoming router's first switch via Ethernet cables + retest Skype. This rules out any WiFi related problems.
So if you're using physical cabling, you must verify the cabling. If you're using any WiFi tech, you must verify your WiFi tech.
Building on the questions asking how the network is laid out... are VLANs being used? (Based on the info laid out, my bet is mo, but figured I would ask anyway)
Who is your SIP provider? How many users are there? Which PBX are you using, and what are the server's specs?
When you say there are problems talking to someone next door, do you mean within the same organization (and on same PBX) or to another organization in a nearby office (and not on same PBX)? The prior means that a problem exists within that office, and absolutely has to get fixed.
Who is your SIP provider? How many users are there? Which PBX are you using, and what are the server's specs?
When you say there are problems talking to someone next door, do you mean within the same organization (and on same PBX) or to another organization in a nearby office (and not on same PBX)? The prior means that a problem exists within that office, and absolutely has to get fixed.
I am not as experienced in networking as some of the others that have posted, but I have managed both hosted and remote VOIP systems.
For testing, do use the speed test in multiple locations. 1)Directly after the modem 2) After the route 3) after the switch and 4) at the same outlet the phone is plugged into. You will see some degradation as you go down the line. But if you have 100Mbs at the modem and 20 at the phone, there is a clue. In addition to the speed test just do a command line ping to your SIP provider and other sites. Do this from each of the 4 locations. If pings are nearing 100 or more that will be another clue.
I found the router was the weakest link in any of the issues I encountered and switching to a better router worked along with correctly setting up QoS. If you have a hosted system, there are going to be some advanced settings on the box that can also make a difference and for that it is best to get your vendor involved if possible.
The complaint that, " some voice conversations are not going well. " does need some more detail as to the exact problem. Are they experiencing dropped calls? Echoing?
For testing, do use the speed test in multiple locations. 1)Directly after the modem 2) After the route 3) after the switch and 4) at the same outlet the phone is plugged into. You will see some degradation as you go down the line. But if you have 100Mbs at the modem and 20 at the phone, there is a clue. In addition to the speed test just do a command line ping to your SIP provider and other sites. Do this from each of the 4 locations. If pings are nearing 100 or more that will be another clue.
I found the router was the weakest link in any of the issues I encountered and switching to a better router worked along with correctly setting up QoS. If you have a hosted system, there are going to be some advanced settings on the box that can also make a difference and for that it is best to get your vendor involved if possible.
The complaint that, " some voice conversations are not going well. " does need some more detail as to the exact problem. Are they experiencing dropped calls? Echoing?
You need to define a few things better:
What is "some conversations are not going well" No sound, metalic sound, clicks ticks,, drops, distortions, one-way sound, ..... disconnects in the midde of a conversation.
The "network is OK"... does that also mean Fit for VOIP? like supporting priorities for RTP, low loss in packet statistics, enough capacity to the outside to have a conversation together with a full speed download?... ( uplink speeds mostly is a limiting thing in those). statistics on packet loss, link loss, capacity etc. are useful. Just "OK" might not cut it. (10 pings in a row without loss)
A network can be Great to transport a few PB of data across the planet and suck for VOIP (in fact such a net work will suck for VOIP if transfers are active).
VOIP (most often) is around two protocols: SIP (for making conenctions) and RTP (actual media transport). SIP should be fast. and not run into dropped packets. it isn't "real time" critical though. (SIP is between Phone and your SIP provider (wether on premises or off premises). SIP packets might become critical if there is too much information in them and Dont fragment is used as well. )
RTP is a different cookie.. That one is real time critical. (it carries voice [ optional video ], in (mostly) 20ms chunks of sound packets that need to travel at constant speed. If late the are dropped, if variance in received packets then distortion will occur. Drops will come up as ticks, clicks or silence.
RTP might go eitehr directly between both sending & listening parties. Or proxies might be in the middle of the conversation. (for example to provide support for NAT or to allow (lawful) intercept).
You can use tcpdump/wireshark/ whatever packet capture tool to make a recording. Wireshark has some tooling on board for analyzing the data. Th RTP stream can also be listened to or analyzed for jitter, etc.
Please provide more info on the exact setup where are the PBX, SBC?, phones relative to the network equipment.
Also things to take into account Ping times to PBX's from the phone connection etc. as well as bandwith specs and how well they are met.
DSL is notorious for loss of packets if things are not done 100% right.
Make diagrams going from one phone to the other with as much detail as possible on things you meet on the way... f.e.
Phone 1 - switch 1 - On premises Router 1 -> ISP1 -> PBX -> ISP2 -> On premises Router 2 -> switch 2 - Phone 2... (if the the switch 1 & 2 are the same then label them as such,,,) Getting to the RTP config might need a packet trace.
Phone 1 - switch 1 - On prem. Rtr 1 -> ISP1 -> PBX1 -> PBX2 -> PBX3 -> ISP2 -> Switch 2 -> phone 2..
etc. etc. and describe the problem you have with each of the scenario's. (or Lack of problems...) and pleas show all your scenarions.... [ if mulitple PBX are in volved then no immediate detail is needed but might help.
What is "some conversations are not going well" No sound, metalic sound, clicks ticks,, drops, distortions, one-way sound, ..... disconnects in the midde of a conversation.
The "network is OK"... does that also mean Fit for VOIP? like supporting priorities for RTP, low loss in packet statistics, enough capacity to the outside to have a conversation together with a full speed download?... ( uplink speeds mostly is a limiting thing in those). statistics on packet loss, link loss, capacity etc. are useful. Just "OK" might not cut it. (10 pings in a row without loss)
A network can be Great to transport a few PB of data across the planet and suck for VOIP (in fact such a net work will suck for VOIP if transfers are active).
VOIP (most often) is around two protocols: SIP (for making conenctions) and RTP (actual media transport). SIP should be fast. and not run into dropped packets. it isn't "real time" critical though. (SIP is between Phone and your SIP provider (wether on premises or off premises). SIP packets might become critical if there is too much information in them and Dont fragment is used as well. )
RTP is a different cookie.. That one is real time critical. (it carries voice [ optional video ], in (mostly) 20ms chunks of sound packets that need to travel at constant speed. If late the are dropped, if variance in received packets then distortion will occur. Drops will come up as ticks, clicks or silence.
RTP might go eitehr directly between both sending & listening parties. Or proxies might be in the middle of the conversation. (for example to provide support for NAT or to allow (lawful) intercept).
You can use tcpdump/wireshark/ whatever packet capture tool to make a recording. Wireshark has some tooling on board for analyzing the data. Th RTP stream can also be listened to or analyzed for jitter, etc.
Please provide more info on the exact setup where are the PBX, SBC?, phones relative to the network equipment.
Also things to take into account Ping times to PBX's from the phone connection etc. as well as bandwith specs and how well they are met.
DSL is notorious for loss of packets if things are not done 100% right.
Make diagrams going from one phone to the other with as much detail as possible on things you meet on the way... f.e.
Phone 1 - switch 1 - On premises Router 1 -> ISP1 -> PBX -> ISP2 -> On premises Router 2 -> switch 2 - Phone 2... (if the the switch 1 & 2 are the same then label them as such,,,) Getting to the RTP config might need a packet trace.
Phone 1 - switch 1 - On prem. Rtr 1 -> ISP1 -> PBX1 -> PBX2 -> PBX3 -> ISP2 -> Switch 2 -> phone 2..
etc. etc. and describe the problem you have with each of the scenario's. (or Lack of problems...) and pleas show all your scenarions.... [ if mulitple PBX are in volved then no immediate detail is needed but might help.