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SHARKFEST ‘10 | Stanford University | June 14–17, 2010 Expose VoIP Problems With Wireshark June 15, 2010 Sean Walberg Vantage Media SHARKFEST ‘10 Stanford University June 14-17, 2010
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SHARKFEST ‘10 | Stanford University | June 14–17, 2010 VoIP is just another application
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SHARKFEST ‘10 | Stanford University | June 14–17, 2010 (but it has special requirements)
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SHARKFEST ‘10 | Stanford University | June 14–17, 2010 Without tools, VoIP is a black box
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SHARKFEST ‘10 | Stanford University | June 14–17, 2010 About Me
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SHARKFEST ‘10 | Stanford University | June 14–17, 2010 The Agenda 1.About VoIP 2.Capturing VoIP 3.Analyzing Signaling 4.Analyzing RTP
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About VoIP Capturing VoIP Signaling RTP
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SHARKFEST ‘10 | Stanford University | June 14–17, 2010 The old way Local Loop
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SHARKFEST ‘10 | Stanford University | June 14–17, 2010 The old way Off Hook Dialtone
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SHARKFEST ‘10 | Stanford University | June 14–17, 2010 The old way Dialing Digits
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SHARKFEST ‘10 | Stanford University | June 14–17, 2010 The old way RING – 90v@20Hz
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SHARKFEST ‘10 | Stanford University | June 14–17, 2010 The old way
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SHARKFEST ‘10 | Stanford University | June 14–17, 2010 The VoIP way I’m calling x1234
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SHARKFEST ‘10 | Stanford University | June 14–17, 2010 The VoIP way Hey, 1234, you’re being called
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SHARKFEST ‘10 | Stanford University | June 14–17, 2010 The VoIP way Use x.x.x.x:xxxx Use y.y.y.y:yyyy
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SHARKFEST ‘10 | Stanford University | June 14–17, 2010 The VoIP way ZZZZZZ
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SHARKFEST ‘10 | Stanford University | June 14–17, 2010 So there are two parts to VoIP Signaling – SIP – H.323 – MGCP – SCCP – Proprietary Voice (Bearer) – RTP (G.711, G.722, G.729a,…)
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SHARKFEST ‘10 | Stanford University | June 14–17, 2010 (two and a half, really) Touch Tones are a problem unto themselves 3212333222333 3212333322321
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SHARKFEST ‘10 | Stanford University | June 14–17, 2010 Network Conditions Affecting VoIP
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SHARKFEST ‘10 | Stanford University | June 14–17, 2010 Loss
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SHARKFEST ‘10 | Stanford University | June 14–17, 2010 Delay
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SHARKFEST ‘10 | Stanford University | June 14–17, 2010 Jitter
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SHARKFEST ‘10 | Stanford University | June 14–17, 2010 Jitter != Delay Jitter Delay Loss (This is from a program called smokeping)
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SHARKFEST '09 | Stanford University | June 15–18, 2009 10, 10, 10, 10 Latency, no jitter 10, 11, 12, 11, 9, 10 Latency and jitter
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About VoIP Capturing VoIP Signaling RTP
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SHARKFEST ‘10 | Stanford University | June 14–17, 2010 Location, Location, Location
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SHARKFEST ‘10 | Stanford University | June 14–17, 2010 Just a simple network
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SHARKFEST ‘10 | Stanford University | June 14–17, 2010 The signaling traffic takes a different path from the RTP traffic
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SHARKFEST ‘10 | Stanford University | June 14–17, 2010 Or, it might do this
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SHARKFEST ‘10 | Stanford University | June 14–17, 2010 Same conversation, different perspectives Here you see inbound latency and jitter, but nothing on the outbound
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SHARKFEST ‘10 | Stanford University | June 14–17, 2010 NAT changes the address Src=A Dst=B Src=C Dst=D The address changes within the cloud!
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SHARKFEST ‘10 | Stanford University | June 14–17, 2010 Set your capture filters
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SHARKFEST ‘10 | Stanford University | June 14–17, 2010 The Packet List window
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SHARKFEST ‘10 | Stanford University | June 14–17, 2010 Summaries are displayed here
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SHARKFEST ‘10 | Stanford University | June 14–17, 2010 By the way… If the signaling or the voice is encrypted, you won’t be able to decode it. Sorry.
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SHARKFEST ‘10 | Stanford University | June 14–17, 2010 Quality of Service for VoIP networks
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SHARKFEST ‘10 | Stanford University | June 14–17, 2010 Add a column for DSCP Edit -> Preferences User Interface->Columns Signaling Tagged RTP Untagged RTP
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SHARKFEST ‘10 | Stanford University | June 14–17, 2010 Are you running a proprietary PBX? Edit -> Properties, Protocols -> RTP
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About VoIP Capturing VoIP Signaling RTP
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SHARKFEST ‘10 | Stanford University | June 14–17, 2010 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
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SHARKFEST ‘10 | Stanford University | June 14–17, 2010 Use the Packet Details pane to see what’s inside the packet
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SHARKFEST ‘10 | Stanford University | June 14–17, 2010 Back to Loss, Delay, and Jitter Jitter is usually a non-issue Delay, within reason, is OK – Clustering/Specific applications notwithstanding Loss isn’t great – TCP retransmits at layer 4 – UDP retries at layer 7
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SHARKFEST ‘10 | Stanford University | June 14–17, 2010 Demos
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About VoIP Capturing VoIP Signaling RTP
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SHARKFEST ‘10 | Stanford University | June 14–17, 2010 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
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SHARKFEST ‘10 | Stanford University | June 14–17, 2010 DTMF Compressing DTMF is bad So many different ways to carry the digits out of band, look for them in traces (see demo)
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SHARKFEST ‘10 | Stanford University | June 14–17, 2010 Three factors that affect voice quality Latency <= 150ms (one way) Jitter <= 20ms Packet loss <= 0.1%
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SHARKFEST ‘10 | Stanford University | June 14–17, 2010 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 Transcoding delay
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SHARKFEST ‘10 | Stanford University | June 14–17, 2010 Packet Loss <= 0.1% Hi Bo *POP* How *POP*e you? Hi Bo How you?
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SHARKFEST ‘10 | Stanford University | June 14–17, 2010 Jitter <= 20ms Better late than never? No. May as well be lost.
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SHARKFEST ‘10 | Stanford University | June 14–17, 2010 Demos
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SHARKFEST ‘10 | Stanford University | June 14–17, 2010 Thanks! sean@ertw.com This presentation will be downloadable from the Sharkfest website.
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