Hi! In my aplication I have a stream created with pjsip. I create the stream as receiver and I have as sender 2 possible streams: another pjsip stream, or a stream created by gstreamer. The configuration from the jitter buffer is the defaul: jitter buffer adaptive, jb=0/1/40 If the sender is the pjsip stream everything works perfectly, but if the sender is gstreamer there is a lot of frame lost, and the audio quality is very poor. Using the Codec L16 with 16kHz I see in wireshark that the RTP packet lenght is allways constant (702 bytes) when the sender is the pjsip stream, but in the case of the gstreamer sender is not constant, the lenght is 1442bytes and 714bytes consecutively. Is there a better way to configure the jitter buffer to work with this, or should I better try the gstreamer sender to generate packets with the same lenght? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20150724/4dc83a03/attachment.html>
