QoS for VoIP On mweb

Zapo

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Is there any QoS / TOS settings that I can configure in my router and pbx to enable better quality when using trunk lines on the internet?

I don't know where one would find such settings/parameters.

I am looking specifically for mweb as the line is a mweb line.

Thanks
 
What router have you got?

I assume you are running voice and data on the same line? Have you considered a separate line just for voice?
 
mikrotik crs125-24g-1s-rm


I am running on its own line, but it does fall back to one line for fail over.

However I was under the impression that but marking my packets with the correct ToS values mwebs network infrastructure would prioritise the voip packets over other data being send on thier network. Not just the data on my line.
 
mikrotik crs125-24g-1s-rm


I am running on its own line, but it does fall back to one line for fail over.

However I was under the impression that but marking my packets with the correct ToS values mwebs network infrastructure would prioritise the voip packets over other data being send on thier network. Not just the data on my line.

I am assuming you running DSL ?
 
Yes, Sorry I forgot to mention the obvious. My bad.

Running 4Meg DSL for the internet line,
2Meg DSL for the sip trunks
3GB as backup
 
I have had very little success using DSL for VoIP, just too darn inconsistent :-( So much time was wasted just monitoring and trying to figure out why it is not working :-( If you are looking at Qos it is most likely that you have a problem of some sort? You getting choppy calls ?
 
Not necessarily, I just want to achieve the best quality posible.

Sadly its for a client that I remotely monitor, So I haven't actually had much time on the phone. But they have had the odd complaint about "bad quality" but this is with sharing data and VoIP on same line.

I have only recently got the dedicated line, I still need to split the data between the two lines.
 
Not necessarily, I just want to achieve the best quality posible.

Sadly its for a client that I remotely monitor, So I haven't actually had much time on the phone. But they have had the odd complaint about "bad quality" but this is with sharing data and VoIP on same line.

I have only recently got the dedicated line, I still need to split the data between the two lines.

Someone correct me if I am wrong...My understanding is that the ToS bits get ignored over the Telkom DSL network and are of little use ? Would be interesting to find out if they do actually make it to the MWEB IPC and are used.
 
If you want to run voice and data on the same line you can
1) Look at a vibe solution. This requires the VoIP provider to support vibe on their side. It will identify voice and prioritise it over data for you.
2) If you have mikrotik you can do packet marking and setup QoS with queues. There are some good articles on the mikrotik wiki.
What neither of the above solutions will save you from is a poor ADSL line. If you have a congested exchange etc. no amount of QoS is going to improve your voice.

If you already have a secondary line and it's stable for the most part, then the client could just be aware that in a fail-over state they could have bad calls if they saturate the line.
 
Yes, Sorry I forgot to mention the obvious. My bad.

Running 4Meg DSL for the internet line,
2Meg DSL for the sip trunks
3GB as backup

Got almost exactly the same setup running at a client - less the 3g backup, using Mikrotik RB2011UAS, also Mweb / Mweb Talk.

If the 2mbit line fails, voip falls over to 4mbit line and enables a simple queue that limits non-voip traffic download to 2mbit and upload to 128kbit and sends a notification email.

No fail-over if only the 4mbit dies.

Running that for more than 1 year with no quality complaints so far (touch wood). Haven't had the need to implement QoS on the router with the current setup.

So can't comment regarding the treatment of marked packets on the ISP's side. Agree with Clarotech - it is best to use queues in Mikrotik to achieve QoS.

If there is exchange congestion or bad copper in the area good QoS will be near impossible to achieve. At another site I have had great results trunking over 3g (telkom mobile).
 
I agree with that,Thanks for your feed back.

However I only see the need for QoS if I am really banging the line and sharing data, I have a dedicated 2meg line purely for a sip trunk with maybe max 3 concurrent phone calls. The QoS will only help me when it fails over, as setup in your case. I was hoping that TOS would help once it is in mwebs data centre. if thier switches are laoded, which is more than likely the case. It might help with some jitter.

But I agree, if you have a busy exchange there is nothing you can do. As far as I understand the PPPoE connection acts as a tunnle from your router to mwebs PPPoE server. Between that is basically a "Tunnel" . but once it hits mwebs internal network, I assume it is just a normal IP network.
 
AFAIK they already prioritise the traffic on their network. For eg. to their voip server, ICMP is de-prioritised over sip traffic.

In a fail-over situation: even if your marks make it through and are recognised, the jitter would surely be caused by bandwidth saturation on the clients side of the DSL? marking packets for the ISP is not going to help that scenario.

IMHO it's probably a waste of some extra bytes.
 
My 1st thought after reading you exchanges was just to kill the thread and say don't use DSL for VoIP but i also know that there is a huge hole in the market for last mile for the micro SME , so let’s try and clear up a few points that are mentioned here.

1/ How good is a DSL line ? I have no idea! They are all different and this is controlled by a variety of factors , distance from the exchange, DSLAM contention, SNR ratio, line attenuation , copper quality, lack of available twisted pairs, the list goes on !

2/ Voice and data on the same line ? This is always a bad idea, all the QoS, ToS , DSCP markings are of limited use if the ISP is not looking for them , they have some but little value client side .Some guys rave about the use of an IPC account, a tunnel directly into the provider, set up on router side as PPPoE with multiple session routing. Sounds good hey ? Data from one ISP and a VPN into another provider, much easier to control the traffic, however, to achieve this you will have to use something like L2TP or IPsec tunnels, the problem is here that the SIP overhead is massive, can be 50% higher using a VPN , since our problem here is bandwidth starvation , we are not in the right direction,
OK, ViBE it was mentioned , ViBE uses byte level QoS and is the best ever QoS on Voice and data i have ever seen ! Alas a small hurdle exists , the voice and data have to be inside the ViBE tunnel for the QoS to work. Most VoIP providers are not ISP's so you have to run your data outside the tunnel and the QoS dont work. MWeb is one of the providers that can give you both voice and data but they do not support ViBe as an ISP.

3/ Dedicated the way to go ? Yes, yes, yes ! So lets assume we have a reasonably stable line ( because we read too much Alice in Wonderland) you should be ok for a few concurrent calls on a good day ! If you ViBe the connection it will be less acceptable to bandwidth starvation, IAX2 will achieve the same thing but is only valid for providers with Asterisk core networks .

4/ 3G failover , Brilliant ! Now we have redundancy ! used to think this was a great idea but lets look at the reality. 90% of tales of wow on DSL are about quality, not my DSL is down, somebody stole the cable ect, if they are complaining about quality it means the router is still synced to The DSLAM and it will not fall over to the 3G, also 3G has issues with restricted APN’s and can be a long task to get the associated provider to un-restrict them.

So in conclusion you see im not the biggest fan of DSL, try and avoid it at all costs , I hope my 2 cents worth has at least made you aware of the challenges .

Go the fibre route, its not as expensive as you think and is the perfect long term solution with increasing broadband demand and quality voice requirements .

PM me if you need more info.

Paul Colmer
 
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