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VOIP bandwidth calculation

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siki85

Technical User
Mar 17, 2009
155
PA
Hi guys,

I ´m a little confused with bandwidth concept for VoIP.

Ok, I´m trying to figure out why we have troubles with cutted-voice on our IP Phones. We have a Siemens Hipath 2036, and according to configuration, the branch is configured with G.711.

Theory says that codec G.711 requires 64kbps, so if have 4 IP Phones, should we require a total bandwidth of 4 x 64 kbps = 256 kbps ? Ok it could be a little more adding overhead and other headers, so the total could be less than 512 kbps.

I also tried to configure G.729, and have activated the VAD (voice activity for silence suppression) but the problem is still appearing.

Am I wrong with this concept ?

Please any advice.....
 
cutted voice" sounds like the sort of thing you can get if you have a duplex mis-match on one of your network interfaces. Check all your relevant network interfaces for errors. Especially the ones connecting your voice servers/gateways/etc... to the network.
 
does the issue with choppy voice happen on internal calls or calls to the outside?
A g711 call requires 128kbps per call total not 64. Overhead is also added to that, but that depends on the connection.
 
Do you have a voice VLAN and QoS on the network? You do not calculate by device, you calculate by call volume, how many active calls are on the network. And if you have Music on hold, there are additional calculations for the media stream for hold music. What kind of Total Network traffic is going on? Have you done a traffic study?
 
Ok, first thank you guys for your responses.

We have several branches pointing to main office. Each branch has a Hipath 2036 while the main site has a Hipath 3800 system.

The cutted voice problem arise when calls are establishes between sites. Our carrier provides us point-to-point links between each branch and main site.

We have 3 subnets (Data, voice and video) for each branch. Those subnets are delivered using 1 fiber optic cable. I use internal vlans on our LAN switches and I suppose that our carrier use trunking through the fiber. I calculated the voice bandwidth considering just 64 kbps per call.

So, should I consider more bandwidth for the voice subnet ?
 
You are not telling us what kind of bandwidth you have between sites or the volume of call expected between them. Regardless, your issue is a qos issue between sites. Voice has to be tagged on your edge devices with high priority (dscp ef). Of course that has to be honored by your provider as well otherwise it will not work.
 
We have different bandwidth configurations, all same for each branch:
- Data link ( 1 Mbps )
- Voice Link ( 512 kbps)
- Video link (just 1 camera for each branch) ( 256 kbps )

We don´t have QoS configured by our carrier.
 
Ok so at g711 you can have a maximum of 4 simultaneous calls per branch. That is without counting any overhead. Truly I would say anything over 3 calls you will have voice quality degradation.
You can do g729 and get many more calls across the 512 link though. Are these bandwidth constrictions one single link to each site? if so what is the speed and what type? P2P, MPLS etc.

 
Try this...


Of top of my hard G711 = 80k (inc overheads) / direction, so bi-directional = about 160k

However choppyness is often dropped or out of order packets.



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