i'm working on a call center, whitch use ip phones in local area networks , and a voip gateway to PSTN networks. the connection between the lan ( ip phones ) and the voip gateway use VSAT, i encountred some
Pb with voice quality, sometimes it work well but other times it's worst. i have tried to change the ios version of the voip gateways , but the Pb alaways exist.
N.B the communication use g729, and the DSPs on voip gateways use codec complexity medium.
Well, any quality that you get is going to be a problem because of the delay that you have with VSAT to begin with. You are starting with a 1/2 second delay. What is the bandwidth that you are having? If it is T1 or better, go with G711 so that the codec delay is minimized. Next, make sure that you have implemented Low Latancy Queuing on both ends of the link. Also check for errors on the links. If you are having a lot of errors normally, voice is going to be that much worse.
we have 2Mpbs on vsat connection , for the codec i can't use the g711 codec because the customer is a call center and need a much number of communication ( +60 communication ) over vsat connection ( 2Mpbs). for the LLQ , i had assigned the strict priority to the voice traffic.
We had the same experience with normal VoIP tests over VSAT. We found that adjusting the codec payload and playout-delays in the VoIP dial-peer helped, but not enough to overcome the high delays of VSAT
I recently designed and deployed a new VSAT Voice over IP network using MPLS VPN in the core network. The major reason this succeeded is that we created a static PVC for each MPLS VPN, a distinct PVC for data and voice, but with a twist. I implemented strict priority (LLQ) for the amount of bandwidth required for that site's voice.
For instance, the total VSAT bandwidth was 128k, and the site has a max of 8 concurrent calls. Using G.729a with compressed RTP and VAD, a max of 64k was allocated for voice, so the LLQ was set to 64k. The remaining 64k was used for network overhead and data, with CBWFQ applied for data prioritization.
In this manner, data can _never_ override voice, but it can indeed use more than its 64k allocation if and only if fewer than 8 calls were in session. This permits up to the subscribers allocation to be used for data, but data usage gets squeezed back down as calls go up.
I tested this to validate the above scenario, and it worked flawlessly without voice degradation. It requires careful design, but I assure you it does work as advertised.
By the way, I also validated that this also works at 64k VSAT bandwidth for up to 4 VoIP calls, including analog handsets.
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