Hi All, We have two pjsip endpoints talking to each other through asterisk, in asterisk we had set canreinvite to yes so that the media is not routed through asterisk.We are using ulaw codec for rtp media. we were continuously monitoring the media quality by collecting rtcp statistics by doing dd(dump detailed) command in pjsua, in this I am consistently seeing RX MAX-Jitter as 1000ms for all calls.I am wondering if there may be some issue with thread locking or systems timer or interrupt in pjsip that sometimes occurs resulting in the 1000 ms addition.And also RTT values varies between 2-3ms as we might expect on a LAN but sometimes jumps to 1000ms.I couldn't understand why the Rx- Max-Jitter value always stays at 1000ms, Is this anything to do with codec part?.please help me in this. Thannks in Advance with regards raja
