Hi, 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?? This is causing problems with the Sipura SPA2000 on our network. The SPA does not like this and is treating many packets as lost packets (even though an Ethereal RTP Analysis trace shows they were not lost). Regards, Andres http://www.telesip.net _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users