Hello I want to analyze VoIP traffic in order to calculate jitter, packet loss and so on. Then comparing them to the values reported by RTCP. Can I analyze that traffic if the application (for example WebRTPC or Hangouts)encrypts the traffic? I guess I can't. Is there an application I can use to analyze this traffic, establishing a connection between two computers I have? asked 18 May '16, 07:35  Haumea |
One Answer:
Hard to say exactly without seeing the capture. To be able to use the jitter and loss analysis which Wireshark offers for RTP, you need access to the information in the RTP header (sequence numbers and timestamps). For SRTP this is still possible as only the audio payload is encrypted there. If some kind of UDP transport layer security is used (but where would you get access to the RTCP in such case?), it may still be possible to evaluate jitter if the codec uses fixed packet rate, and loss if the jitter is small. With big jitter and no access to sequence numbers, detection of loss would be more complex (counting packets over a time interval leaves some uncertainty). With a TCP-based encrypted transport, loss at packet level can be seen directly on TCP level but does not exist at all from application point of view; on the other hand, the jitter coming from eventual retransmissions caused by packet loss is so massive that the "natural" one becomes insignificant. But also here, the packet rate must be constant so that jitter could be evaluated without access to the timestamps (unless TCP timestamp option is used). answered 18 May '16, 07:58  sindy |
