Hi Tim I will try to give you some suggestions from the top of my head. 1. As a purely static approach you can estimate this by looking at the playback and recording buffer sizes. Adding receiver jitter buffer medium values and expected NW transport time, it should give you a theoretical idea. It is also good to get familiar with these params since if latency is a key thing for you you might want to optimize these buffers later on to make them fit your use case and network conditions. 2. If you are not looking for auto generated statistics a very simple approach is just to use Audacity and put the two phones in the same room and mute one of them. Make a noise spike. You will then see this spike in audacity and you will also see the corresponding spike when its played back in the receiving phone. After that you can just manually measure the time between the 2 spikes and you will get an idea on the latency. Its good for empirical testing if you do not need automation. 3. Maybe there are already some good tools in pjsip that I have not tried but you should be able to do a lot looking at for instance RTP sequence numbers. If the timing is correct between the two devices I think you can quite easily parse a log file to see when timestamp A was received in B. However doing this you have to take into consideration when in the audio flow you are looking at the timestamp since the buffers will affect this value as well. If you just want to look at network latency Wireshark with the clients in the same computer should also give you an idea. There are probably other more sophisticated ways to do this as well. /Johan ________________________________ Este mensaje se dirige exclusivamente a su destinatario. Puede consultar nuestra pol?tica de env?o y recepci?n de correo electr?nico en el enlace situado m?s abajo. This message is intended exclusively for its addressee. We only send and receive email on the basis of the terms set out at: http://www.tid.es/ES/PAGINAS/disclaimer.aspx
