I have a question regarding the Asterisk Packet Time for SIP Calls. It is hardcoded at 20ms but when I do an RTP Analysis on a stream it is clear that these packets are not spaced out at 20ms. In general you see something like:
Packet 50 - Delay 50ms Packet 51 - Delay 5ms Packet 52 - Delay 5ms Packet 53 - Delay 50ms Packet 54 - Delay 5ms Packet 55 - Delay 5ms
Is there anyway to space them out evenly at 20ms??
The 20 ms is not the inter-packet timing, its the relative content of what's within the packet. In other words, the packet contains 20ms of encoded voice.
If the inter-packet times (delays) are large, as they would seem to be in your example, then something else is not right. Possibly a half-duplex ethernet connection, something else running on the server, router buffers, etc.
On a typical * --> C7960 local call, I generally see from 1ms to 20ms inter-packet delays. Seldom (if ever) anything above 20ms.
I gather from your reply that there are recommendations regarding the ethernet connection on your Asterisk server? half-duplex seems bad. Could you elaborate a bit on that?
/Olle
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