VoIP jitter

Before I googel around. I know that from Cisco, an acceptable VoIP deplay is 150ms one-way. I'd like to get some feedback on what is an acceptable jitter time for VoIP. Is it depending on the codec? Is it depending on the network devices?

I have a Cisco switches infrastructure and I have NEC VoIP phone.

Thx
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leblancAccountingAsked:
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James HIT DirectorCommented:
It is dependent on the codec. G711 and G729 cannot handle packet loss and delay needs to be less that 100ms.
On network devices, you configure them for QoS and mark the VOIP traffic and give it priority over the rest. It also depends where you are connecting to, over a WAN or internet?
Either way you need to prioritize your traffic and ensure that VOIP is always given greater priority.
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leblancAccountingAuthor Commented:
I am testing the jitter over the MPLS in preparation for the VoIP deployment. I am using iperf but I am not sure the paramaters I should use for UDP with the corresponding codec.

What is the recommended jitter for g711 or g729? is it 2ms? 10ms? 17ms?
 
How about the buffer for UDP for voice?

How big is the packet size for g711 and g729?

Thx
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James HIT DirectorCommented:
OK. This is where you will have to ask you MPLS provider IF they are applying QoS to your circuit and what policy are they using. From there you will have to match those settings on your router. This needs to be an exact match for QoS to properly work.
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leblancAccountingAuthor Commented:
no I understand, They are applying QoS on their circuit.
I am injecting packets with iperf to test my jitter. My question is not about QoS. It is about jitter. What is a reasonable response time for VoIP jitter? and how big is the packet size for g711 and g729?
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pergrCommented:
Latency and jitter is different things.

There is really no problem with latency (more than 100ms its fine) - you will just have a delay when talking to each other.

Jitter is the difference in latency between different packets in the same flow (call). The limits depends on the phones. Basically the phone will not play the sound immediately when a packet is received - it waits a bit in case the next packet will be extra delayed.  That time is called jitter buffer. It is best to use dynamic jitter buffers,  if your phone supports it. It means that when the network is bad the phone automatically increases the jitter buffer a bit.

Any packet that arrives after the jitter buffer time is expired will not be played, and there is a gap in the call.
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leblancAccountingAuthor Commented:
So what is a reasonable response time for VoIP jitter? I am using iperf to get some stats and I have in average 6ms for the jitter. I am simulate UDP traffic with 64kbps packet size. But I am not sure if 6ms is a reasonable time for jitter.
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pergrCommented:
Anything under 30ms should be fine. Some phones might be able to handle up to 100ms jitter.

When you run the test, make sure that you set the TOS marking of the packets to the same as your phones TOS marking.
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leblancAccountingAuthor Commented:
"Anything under 30ms should be fine. " Where can I validate this statement? Thx
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leblancAccountingAuthor Commented:
That is exactly what I am looking for http://www.ciscopress.com/articles/article.asp?p=357102. Thank you very much.
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