VOIP QoS Setup

[Peon]

Hi peeps,

Selling VOIP to customers can be tricky, especially when telkon and other major providers have outages.

Most clients or small businesses run ADSL and use VOIP. More often than not one worker is sneaking utorrent or someone is addicted to youtube thereby hogging all the bandwidth. VoIP calls then suffer and its the providers fault. Easily rectified with a QoS rule. So the hope of this thread is to explore the QOS options and provide a reference point for others who want to setup a QOS rule. Below is a screenshot of a typical QOS setup page. Could one of the admins please embed the screenshot properly so it doesnt open in a new tab.



There a whole bunch of things going on here. All the selections up until choosing the interface is pretty straight forward.

What does one make of things like DSCP and IP Precedence, TOS and 802.1p?

Like I alreadfy mentioned, hopefully this thread can discuss this topic and expand on the importance of QOS'ing VOIP traffic to improve VOIP calls which I feel personally alot of admins and technicians overlook on client ADSL networks and setups.

 

[tRoN]

This is a great idea. Hopefully someone will help.

Do all router models have the same type of settings.

 

[Peon]

Things like DSCD and IP Precedence are general networking concepts.

I was/hoping some VOIP gurus could share their knowledge on and understanding on QOS'ing voip traffic, effect of codecs etc,etc.

Im pretty sure of you were to survey 1000 small businesses around the country and ask what internet they have, most would be 4 MB. But company owners dont understand that that is the download, not upload and upload is very important to call quality and there is only 50KB/s of it - at best of times on a perfect connection.

One will have to balance act rules to optimize performance while still maintain quality, ie: "Hello im struggling to hear you, its a bit soft". As I understand it some admins swear on dedicating at least 40KB/s to VOIP traffic, hopefully more can comment on this.

 

[morkhans]

As a rule I would never mix voice and data. ADSL is cheap enough for a small business to be able to dedicate a line to VoIP only. Less pain, less policing required.

What you should also be looking for in a VoIP provider if you are going the ADSL route is one that can offer vibe support which is a relatively inexpensive QoS system purpose built for VoIP.

 

[BitCo]

Hi Peon,

I agree with Morkhans and try not mix Voice & Data over a single DSL line.

We recently had a customer who had a small home office and was only able to get a single ADSL line as the Telkom exchange in the area was full, as they wanted to use a hosted PBX platform to keep costs down we needed to do Voice and Internet for 4 users over a single line.

We managed to do some shaping on the router to limit the amount of internet access & allow voice services but it doesn't really work as often the customer experience breaking up on the calls.

We haven't had much experience with Vibe but i have heard good things and its worth you checking it out.

Regards,

BitCo

 

[Peon]

Interesting BitCo, you use past tense in your post, client moved on?

Where they using 4mb or 1mb? Telkom ISP? Could the congested exchange not account for poor performance? Should G729 not assist in these issues? Dedicated hardware PBX wants 38.4Kbps?

Could we agree that upload is a significant problem with voip on DSL lines?

 

[tRoN]

Hi Peon,

I agree with Morkhans and try not mix Voice & Data over a single DSL line.

We recently had a customer who had a small home office and was only able to get a single ADSL line as the Telkom exchange in the area was full, as they wanted to use a hosted PBX platform to keep costs down we needed to do Voice and Internet for 4 users over a single line.

We managed to do some shaping on the router to limit the amount of internet access & allow voice services but it doesn't really work as often the customer experience breaking up on the calls.

We haven't had much experience with Vibe but i have heard good things and its worth you checking it out.

Regards,

BitCo

What would have been the least cost effective line alternative for this individual?
In other words what was his solution to the problem at the end of the day?

 

[TheGuy]

DSCP marking and most of the other stuff are not going to work as you need your upstream provider to listen to it but they don't.

The big problem with Qos is you have hardly any control over incoming bandwidth. There is a way you can control incoming TCP traffic but as far as I know there is no way to control UDP traffic.

The best way to limit P2P traffic is to block it completely. Only allow outgoing ports that are required. Then also throw in some monitoring to see how much bandwidth people use.

 

[BitCo]

Hi Guys,

Yea I was speaking in past tense as the solution was like that for only about a month. We ended up moving the internet browsing onto a separate 3G router (Netgear 3G) as the ADSL was dead slow and constantly caused issues with the Voice.

We continued to run the Voice over the Dedicated ADSL and we haven't had any problems since.

In general we still do a lot of Voice services over ADSL Lines, we probably do about 500 concurrent calls at any time on our network originating from customers using SIP / IAX over a dedicated ADSL Line.

1st prize is always to have a guaranteed last mile connection that is uncontended and synchronous but you can’t always get it & generally very expensive.

ADSL there are certain factors we look at when connecting a client onto our Voice Network

1: Latency on line (Needs to be under 50ms and needs to be consistent)
2: Packet Loss (Packet loss causes problems so we always test to ensure no packet loss)
3: ISP Account (We have a @bitco ISP account which we give to clients free of charge, this is a SAIX based account which allows low latency to our network)
4: Line Speed (we are generally always limited by 512kbps upload but it’s always best to test this as sometimes is far less)

As a rule here at BitCo we do 15 Concurrent calls over a DSL Line (512Kbps Upload) when connecting to one of our PBX systems as we use IAX2.

If a customer is using our virtual PBX platform over DSL and using straight SIP (G729) we limit it to 8 concurrent calls.

Here is a bandwidth calculator which can give you an idea of what the different codecs use - http://www.asteriskguru.com/tools/bandwidth_calculator.php

I hope some of my info was helpful

Regards,

BitCo

 

[fixx]

You should try to physically split VoIP and Internet data. VoIP has a hard time dealing with variable latency rather than high latency. Asterisk has measures to compensate for this, but its only effective up to a certain point.

 

[web_dude]

I use a diginet for the voice traffic. Is point to point (from my premises to the VOIP supplier) and as it run on the local loop is quite cheap (I think under R1000) and it can easily take 16 simultaneous calls.

Good thing as well that the diginet is managed by my supplier, which means I don't have to deal with Telkom.

If the Diginet goes down, the voice traffic is moved to my Internet connection (which is wireless with syncronous speed). And my firewall is automatically set to give priority to sip traffic.