Hello every one…
just i want to know about rtp qos. my question is when we make call from server-A to local server-B working perfect like audio quality very good … some time internet speed going down like very slow then call quality going very poor incase drooping rtp packets…
my question we can set some values in dial plan for don,t send rtp packet forwarding which is voice break like wrong rtp packet ? or some rtp filter we can use for skipping Brocken packets
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What you are likely to be hearing is the result of not receiving RTP packets or receiving them too late. That’s equivalent to not sending them, so I think your proposed solution won’t work.
The best you can hope for is for the end system to fill in based on the last good RTP frame and that is best left to the end system.
You mentioned QOS. QOS refers to marking RTP packets, typically to avoid being delayed, but that depends on the network honouring those markings, which is only likely to happen within your LAN. It also assumes that the amount of prioritised traffic is small enough that there is enough other traffic left to be downgraded to make way for it.
In reality you need to address the network problems, which may mean moving to a more upmarket ISP, or moving, from the cloud, to a local server. I never really understood why so many people want to operate PABXes in the cloud.
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