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QoS for Voice over bonded T1's

Last Modified: 2012-08-13
Ok I have two point to point T1's that I plan on bonding together vial Multilink1/ppp multilink group 1. The problem is I have voice and data going down these ciruits. Before the bonding we had one of the T's allocated with only 6 channels for voice and the rest for data. Now I'd like to configure QoS to do this so that voice doesn't end up flooding the channels. How can I do this?
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I forgot to mention that we don't use a separate vlan for voice and it's a Seamens PBX (POTS)


I would like for the voice to never exceed 384 k/bits (6 channels)
I'm not a CCIE but I think there may be an issue here.

You have half channels voice, and half data. If you bond to internet t1's that implies to me that they will all be data.
This in tern implies to me that you would be doing Voice Over IP.
If you are not doing voice over IP, than you cannot bond the T1's because you need voice channles on  your T1, which is different than running data over PPP.
I spend most of my time in switches and with Frame, so I apoloigze if I'm missing something here.

If you are doing voice over IP, than QOS should work great. But the type you apply depends on the type of hardare you have.

Traffic shapping using policing and classifiing traffic using DSCP is what you would do for VOIP.

It sounds like you also need MLPPP

here are some links, this is a for QOS on a 4507, which is what I do all my traffic shaping on, but I found it is helpful in general:

Also, policy based routing is part of most qos configs:

I am a bit confused on your install... You have a bonded T1 giving you a total of 3MB... I am assuming before you went with VoIP sulution you deticated 6 channels for POTS lines... but than you bought a VoIP solution? Since you have a POTS PBX.

VoIP can not flood a channel... actually you are woring about the inverse... data will burst to its maximum transfer rate and than back down... A VoIP call selects a codec and than sends a unicast RTP media stream at a CBR Constant Bit Rate or less... So depending if you compress the codec or not here is how you would calculate:

G.711u (Which is uncompressed) is 70kbs or so... but, using  PPP over a multilink HDLC circut looses some framer packets and is more like 82... but... I calculated my stuff using 76... to take a mean...

384 divided by 76 is about 5 calls... now you can compress the call ... and fit it in the 384 or increase the amount you alot to VoIP...

One thing you maybe forggeting with IP you do not allocate the bandwidth... you prioritize it... so you would only be using 76kbs per call going on at that time... so if you do not have 100% usage... you are actually using a lot less for voice most of the time...

Diffserv works well over PPP... tag the VoIP packets by the device or an application layer filter and you should be just fine...

Good luck
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