VOIP how to calculate RTP packages delay

I am making a VOIP application for m开发者_运维百科obile platform. My question is what algorithms should be used to calculate whether the RTP package is "expired". I transfer PCMU encoded audio wrapped into RTP packages via UDP. As you know some of the datagrams are not delivered while others are delivered late and its pointless to play the audio in those packages. Using the sequence number in the RTP header you can calculate the lost packages, but I want to know how to calculate when a packet is late. I saw that there is something called jitter which basically measures the difference between the times for sending two consecutive packages and receiving them. Can I use that? Or something else?

What your application considers 'expired' or 'too late' is really up to your application, but you should make sure you play out the audio evenly. So the measure for too late is the size of your playout buffer and the size of the buffer depends on the type of application. Bidirectional communication will need a smaller buffer than simple movie playback.