On Monday 22 December 2003 15:36, Rich Adamson wrote: > > 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. Thanks for your Input Rich. I went ahead and tested this on our production servers and sure enough the inter-packet times are 20ms. There must be something happening with our LAB Asterisk. It could be the CBQ traffic shaping software we have running on it. I will fiddle around with it to see if it changes anything.
Thanks! Andres > > Rich > > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
