Hi! Am 12.05.2010 06:37, schrieb Sevana Oy: >> Does somebody know how pjsip writes the wavefile? Will it be written >> exactly like to the audio device (with possible jitter buffer >> under/overrun and playback-speed adjustments) or will the voice sample >> be written just one after the other to the wave file? > > Call audio will be saved one after another into the same file, however, > this can also be solved in order to receive recording of a single call. That's not what I asked - maybe I should make myself more clear: During a normal phone call, the receiver may do manipulations to the audio stream before playing back the audio to the user. For example SIP clients often have dynamical jitter buffer - when the buffer gets empty the playback speed will be reduced, when the buffer gets full the playback speed will be increased, old packets may be ignored completely. When a call is recorded, this manipulations are not needed because it doesn't matter if packets arrive late as for recording there are no real-time constraints. The receiver can wait until the RTP packets are received and then it saves the audio payload without need for manipulation to the wav file. If a client writes a wav file like described above, and the is no packet loss - the recorded file will always be identical to the file sent by the other party. If a client does audio manipulation also for recordings, then the recorded file will differ from the original and MOS should be different. regards Klaus
