How much bandwidth should you have for VOIP calls??

[Ralph]

I have heard that you need between 90 - 100k of bandwidth per line.

In testing 4 simultaneous calls they seemed to take up about 175k per call on average and the amount of bandwidth per call decreased as more calls were placed.

Anyone know what a good guideline or rule of thumb for planning VOIP implementations when it comes to bandwidth??

Thanks again

 

[RobLloyd]

3CX says about 80k/call. I don't know if there are official #'s posted somewhere but 60k-80k seems be the average that I've seen. If you plan on 100k then you should be fine.

 

[RobLloyd]

So your 4 calls took 700k? That seems like a lot and would fill most DSL/Cable upload speeds. At least I did think that, I just did a test and got
Download Speed: 12393 kbps
Upload Speed: 1865 kbps
My cable improved 100%!

What did you test it with?

In testing 4 simultaneous calls they seemed to take up about 175k per call on average and the amount of bandwidth per call decreased as more calls were placed.

 

[Ralph]

Woops guess it helps to make sure of your baseline first

I took another look at the server and made sure that only the pbx was running

This time using performace monitor I discovered that I am using 45k of bandwidth for the first call and it appears that each additional call uses 20k of bandwidth so with 3 simultaneous calls I am using 85k of bandwidth. Is this normal?

 

[Fynskisb16]

It really depends on your VoIP provider. Most use G711a/ulaw. Other are using g729a which is much smaller. If your using G711 I'd plan on at least 100k per call. 3cx only supports g711a/u and GSM-FR as far as I can see on my server.

Cisco says this:
http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a0080094ae2.shtml
Codec Information
Bandwidth Calculations

Codec & Bit Rate (Kbps)
Codec Sample Size (Bytes)
Codec Sample Interval (ms)
Mean Opinion Score (MOS)
Voice Payload Size (Bytes)
Voice Payload Size (ms)
Packets Per Second (PPS)
Bandwidth MP or FRF.12 (Kbps)
Bandwidth w/cRTP MP or FRF.12 (Kbps)
Bandwidth Ethernet (Kbps)

G.711 (64 Kbps)
80 Bytes
10 ms
4.1
160 Bytes
20 ms
50
82.8 Kbps
67.6 Kbps
87.2 Kbps

G.729 (8 Kbps)
10 Bytes
10 ms
3.92
20 Bytes
20 ms
50
26.8 Kbps
11.6 Kbps
31.2 Kbps

 

[tjabaut]

From what I have learned it is all about the codecs that you choose.

3CX does not currently support g729 which is the most acceptable bandwidth saving codec in use. g711u/a one voice call sets up two 64 Kbit RTP streams, plus overhead (source: http://voip-info.org/wiki/view/Bandwidth+consumption). Therefore, one call over g711 would require roughly 160kbps, so your test of about 175kbps per call would be accurate.

This is a little exaggerated due to packet transmission/retransmission, overhead, etc.; as well as the fact that since VoIP is not transmitting "white noise you are not truely "full duplex", but you should always figure in/out transmissions in your calculations.

 

[Anonymous]

Ok total amount of calls possible is depending on the BHT.


So lets say you want to have a BHT of 4. A blocking of 0.010.

We use G711 (or PCM) this is typically 64kbps uncomressed. The more advanced phones and ATA allow compression. So lets do some numbers and you see that you need 720kbps to do that, that gives you 10 voice paths and only one in a 100 call rejections.

Lets do 1 call on all compressions. (packet duration by miliseconds and samples)

Milliseconds Samples BW in kbps Voice Paths
20 160 400 5
40 320 360 5
60 480 347 5
80 640 340 5

So having said that your BHT of 4 will require 720kbps on 40 milisec, you see it is not as simple of adding 1+1+1+1 is too much .

Lets take the 40milisec and use RoLloyds download speed (keep in mind you should always take the lowest available speed) we want a blocking of 0.010 we have a 1865kbps bandwidth that means we can have 16 BHT.

Pretty good

So what does this all mean?
BHT is most possible hrs of call traffic during the busiest hour of operation. Lets say concurrent calls (sort of).
Blocking is the failed calls due to nolines available. so 0.010 means 1 in a 100 calls gets dropped.

The voice paths is the number of simultaneously paths that the network is able to transport.

So hope this helps

 

[SY]

for ptime=20 (RTP packet time is 20 ms)
one direction:
GSM - ~13kbps(native)+~11kbps(UDP headers)+~5kbps(RTP headers) =~ 29kbps (~3.5 kB/s)
G.711- 64kbps(native)+~11kbps(UDP headers)+~5kbps(RTP headers) =~80kbps(~10kB/s)

total traffic (sum in both directions)-
GSM 58kbps(7 kilobytes per second)
G.711 - 160kbps (20kilobytes per second)

SPEEX and iLBC - comparable with GSM, but slightly (10-15 percents) more.

 

[Ralph]

Thank you

Thanks for all the information.

Let me take a a little time to digest this :shock:

Take care

 

[Cloverlick]

One key element that seems to be overlooked in this discussion string is the fact that a VoIP-exclusive call requires two (2) connections... each using bandwidth.

With our setup (VoIP exclusive - G.711).... we have an inbound provider connection that averages 60-80k... and an outbound connection that averages 60 - 80k.

Using the highest average variable for our setup...

80k --> 3CX --> 80k to phone ext.

Wherefore...if your receiving ext. is INSIDE your LAN, you would need 80k-ish per active call.

If your receiving ext. is OUTSIDE of your LAN (as all of ours are) you would need 160k per active call.

And remember... bandwidth is not the only consideration....

Latency, hops, bottlenecks and heavy-traffic make VoIP calls very sad.

GC

 

[Anonymous]

One key element that seems to be overlooked in this discussion string is the fact that a VoIP-exclusive call requires two (2) connections... each using bandwidth.

Ehhh, well that depends on the total legs you introduce. Appart from that it also depends on how much talking you do.

Anyway the KEY ELEMENT you are revering to is also know as Voice Paths, and if you check some of the posts you will find that the KEY ELEMENT is not over looked.

The bandwidth is also depending on what your devices can handle, for example 3cx does not do something clever with the compression hence the G.711 is 64kbps uncompressed and your story about 80kbps per voice path is correct. (Actually it is 80kbps, the drop to 60kbps is due to silence on the line so the correct bandwidth with overheads etc is 80kbps).

But if i introduce payload lenght you can bring that right down.

 

[SY]

One key element that seems to be overlooked in this discussion string is the fact that a VoIP-exclusive call requires two (2) connections... each using bandwidth.

With our setup (VoIP exclusive - G.711).... we have an inbound provider connection that averages 60-80k... and an outbound connection that averages 60 - 80k.

Using the highest average variable for our setup...

80k --> 3CX --> 80k to phone ext.

Wherefore...if your receiving ext. is INSIDE your LAN, you would need 80k-ish per active call.

If your receiving ext. is OUTSIDE of your LAN (as all of ours are) you would need 160k per active call.

And remember... bandwidth is not the only consideration....

Latency, hops, bottlenecks and heavy-traffic make VoIP calls very sad.

GC

My description is completely fair, isn't it? What is overlooked there?

for ptime=20 (RTP packet time is 20 ms)
one direction:
GSM - ~13kbps(native)+~11kbps(UDP headers)+~5kbps(RTP headers) =~ 29kbps (~3.5 kB/s)
G.711- 64kbps(native)+~11kbps(UDP headers)+~5kbps(RTP headers) =~80kbps(~10kB/s)

total traffic (sum in both directions)-
GSM 58kbps(7 kilobytes per second)
G.711 - 160kbps (20kilobytes per second)

SPEEX and iLBC - comparable with GSM, but slightly (10-15 percents) more.

 

[Anonymous]

SY,

Your a right.

Not sure what or where cloverlick is comming from but I think he/she just has a bad day.

 

[Ralph]

Ok let me make sure I understand this a little

Ok, so an average phone conversation using G711 with 3CX as the PBX would require 80kbps both directions for a total bandwidth need of 160kbps. A 1.6 meg up and 8 meg down internet connection could then handle 10 simultaneous phone calls assuming that was the only traffic on the network?

Is there a way to brind the bandwidth requirement down without a significant decrease in voice quality?

Thanks again,
Ralph

 

[tjabaut]

Ralph,

It will all depend on the CODEC used. there are numerous VoIP Codecs used today. I believe 3cx uses g711 is because it offers the best voice quality as it is basically using the same mod/demod as a standard 64k voice channel; and 2 because it is FREE and open source. The best low bandwitdh high quality codec is g729, but it requires licensing.

Currently, I do not believe that there is any facility in 3cx to install third party codecs; however is does support g711, and I believe speex and gsm which are lower bandwith, lower quality codecs. you will have to try them for yourself, as your mileage may vary.

I know some other systems acutally use a codec passthru right to the VoIP Phone, as almost all natively support g729. This would not force you to need a seperate license, but that would only pertain to calls that actually terminated in a call (ie answered by the phone). If vm were to answer than you would have to use a codec supported by 3cx.

this can get really drawn out. Does anyone know if 3cx is working on the ability to integrate 3rd party codecs?

 

[RobLloyd]

I hope they are as here in the US, most small businesses has cable or DSL with 384k-768k upload speeds. Add email and all the other internet traffic and there's not much left for voice.
I'm having a hard time trying to get clients to switch to VoIP without upgrading their internet. The cost savings aren't there yet.
My next client will mix both PSTN and VoIP, using VoIP for overflow.

this can get really drawn out. Does anyone know if 3cx is working on the ability to integrate 3rd party codecs?

 

[Anonymous]

My next client will mix both PSTN and VoIP, using VoIP for overflow.

That at this stage is possibly the best way to go about it, have VoIP but we have not reached maturity as yet to have ONLY VoIP.

A 1.6 meg up and 8 meg down internet connection could then handle 10 simultaneous phone calls assuming that was the only traffic on the network?

In your calculations always take the lowest bandwidth, so you can approach this two ways. Saying:
1600kbps how many calls can that handle? 12 calls (representing in 20 voice paths) and having 1 in a 100 calls rejected.

Or

10 calls at the same time how much bandwidth do I need. Same story 10 concurrent calls need 18 voice paths representing in 1440kbps (18 voicepaths).

Is there a way to brind the bandwidth requirement down without a significant decrease in voice quality?

Well sort of, you can but you need some $$$$. You can nominate a bandwidth segment for voice only, this is costly and if you do it static you mis out on the full capacity of the line.
So some smart people came up with TOS and QoS, the QoS will allow you to nominate what will have priority on the network. This means that every VoIP packet gets a tag with URGENT on it (just like you send mail or a parcel) the switch/router will than determine if there is enough bandwidth for the packet to go through if not it will stop lower priority packets and buffers them to allow VoIP to go through.

All that works great, but it will have limitations based upon your hardware. But this is sort of the high level of it.

Hope this helps

 

[SY]

Re: Ok let me make sure I understand this a little

Ok, so an average phone conversation using G711 with 3CX as the PBX would require 80kbps both directions for a total bandwidth need of 160kbps. A 1.6 meg up and 8 meg down internet connection could then handle 10 simultaneous phone calls assuming that was the only traffic on the network?

Is there a way to brind the bandwidth requirement down without a significant decrease in voice quality?

Thanks again,
Ralph

if only voice traffi - then 20 call (upstream usage will be 80 kbps per call)

You can use GSM codec with 3CX PBX (SPEEX and iLBC suported as well). it will reduce requirement for bandwidth up to 2.7 times.

 

[Anonymous]

if only voice traffi - then 20 call (upstream usage will be 80 kbps per call)

Ehhhhh, that is 20 voice paths that represents in 12 calls concurrent. Unless I am really wrong somewhere...... that can happen.

 

[tjabaut]

Henk

Can we get back to some basics for a minute. How is 12 simultaneous calls only 18 voice paths? Shouldn't it be 24 vp's (2 x the # of calls).

I am not sure, just asking.