Presentation on theme: "SHARKFEST '08 | Foothill College | March 31 - April 2, 2008 Exposing VoIP problems with Wireshark April 2, 2008 Sean Walberg Network Guy | Canwest SHARKFEST."— Presentation transcript:
SHARKFEST '08 | Foothill College | March 31 - April 2, 2008 Exposing VoIP problems with Wireshark April 2, 2008 Sean Walberg Network Guy | Canwest SHARKFEST '08 Foothill College March 31 - April 2, 2008
Signaling protocols SIP (from the IETF) H.323 (from the ITU) MGCP IAX SS7 (Telco) GSM (Telco/Cell) SCCP (Cisco Skinny) Vendor specific
The role of signaling Indicate to the remote end that a call is coming Establish the codec to be used for voice Establish the addresses of the endpoints Get out of the way Tear down the connection once it’s done
The properties of RTP RTP simulates the real time voice normally carried over a wire 4KHz voice bandwidth = 8KHz sampling rate (Nyquist) 8 bits/sample * 8KHz = 64,000bps (DS0) A Codec (G.711u/A law, G.729, G.726, etc) Most codecs use 20ms voice samples = 50pps Even with compression, you have a fairly consistent packet rate, only the size changes
Three factors that affect voice quality Latency <= 150ms (one way) Jitter <= 20ms Packet loss <= 0.1%
Latency <= 150ms (one way) Hi, how are you? Hello? Oops, sorry, go ahead Fine, I oh hello, go ahead Path delay Serialization delay Jitter buffer, Transcoding delay
Packet Loss <= 0.1% Hi Bo *POP* How *POP*e you? Hi Bo How you?