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

SHARKFEST ‘10 | Stanford University | June 14–17, 2010 VoIP is just another application

SHARKFEST ‘10 | Stanford University | June 14–17, 2010 (but it has special requirements)

SHARKFEST ‘10 | Stanford University | June 14–17, 2010 Without tools, VoIP is a black box

SHARKFEST ‘10 | Stanford University | June 14–17, 2010 About Me

SHARKFEST ‘10 | Stanford University | June 14–17, 2010 The Agenda 1.About VoIP 2.Capturing VoIP 3.Analyzing Signaling 4.Analyzing RTP

About VoIP Capturing VoIP Signaling RTP

SHARKFEST ‘10 | Stanford University | June 14–17, 2010 The old way Local Loop

SHARKFEST ‘10 | Stanford University | June 14–17, 2010 The old way Off Hook Dialtone

SHARKFEST ‘10 | Stanford University | June 14–17, 2010 The old way Dialing Digits

SHARKFEST ‘10 | Stanford University | June 14–17, 2010 The old way RING –

SHARKFEST ‘10 | Stanford University | June 14–17, 2010 The old way

SHARKFEST ‘10 | Stanford University | June 14–17, 2010 The VoIP way I’m calling x1234

SHARKFEST ‘10 | Stanford University | June 14–17, 2010 The VoIP way Hey, 1234, you’re being called

SHARKFEST ‘10 | Stanford University | June 14–17, 2010 The VoIP way Use x.x.x.x:xxxx Use y.y.y.y:yyyy

SHARKFEST ‘10 | Stanford University | June 14–17, 2010 The VoIP way ZZZZZZ

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,…)

SHARKFEST ‘10 | Stanford University | June 14–17, 2010 (two and a half, really) Touch Tones are a problem unto themselves

SHARKFEST ‘10 | Stanford University | June 14–17, 2010 Network Conditions Affecting VoIP

SHARKFEST ‘10 | Stanford University | June 14–17, 2010 Loss

SHARKFEST ‘10 | Stanford University | June 14–17, 2010 Delay

SHARKFEST ‘10 | Stanford University | June 14–17, 2010 Jitter

SHARKFEST ‘10 | Stanford University | June 14–17, 2010 Jitter != Delay Jitter Delay Loss (This is from a program called smokeping)

SHARKFEST '09 | Stanford University | June 15–18, , 10, 10, 10 Latency, no jitter 10, 11, 12, 11, 9, 10 Latency and jitter

About VoIP Capturing VoIP Signaling RTP

SHARKFEST ‘10 | Stanford University | June 14–17, 2010 Location, Location, Location

SHARKFEST ‘10 | Stanford University | June 14–17, 2010 Just a simple network

SHARKFEST ‘10 | Stanford University | June 14–17, 2010 The signaling traffic takes a different path from the RTP traffic

SHARKFEST ‘10 | Stanford University | June 14–17, 2010 Or, it might do this

SHARKFEST ‘10 | Stanford University | June 14–17, 2010 Same conversation, different perspectives Here you see inbound latency and jitter, but nothing on the outbound

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!

SHARKFEST ‘10 | Stanford University | June 14–17, 2010 Set your capture filters

SHARKFEST ‘10 | Stanford University | June 14–17, 2010 The Packet List window

SHARKFEST ‘10 | Stanford University | June 14–17, 2010 Summaries are displayed here

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.

SHARKFEST ‘10 | Stanford University | June 14–17, 2010 Quality of Service for VoIP networks

SHARKFEST ‘10 | Stanford University | June 14–17, 2010 Add a column for DSCP Edit -> Preferences User Interface->Columns Signaling Tagged RTP Untagged RTP

SHARKFEST ‘10 | Stanford University | June 14–17, 2010 Are you running a proprietary PBX? Edit -> Properties, Protocols -> RTP

About VoIP Capturing VoIP Signaling RTP

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

SHARKFEST ‘10 | Stanford University | June 14–17, 2010 Use the Packet Details pane to see what’s inside the packet

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

SHARKFEST ‘10 | Stanford University | June 14–17, 2010 Demos

About VoIP Capturing VoIP Signaling RTP

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

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)

SHARKFEST ‘10 | Stanford University | June 14–17, 2010 Three factors that affect voice quality Latency <= 150ms (one way) Jitter <= 20ms Packet loss <= 0.1%

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

SHARKFEST ‘10 | Stanford University | June 14–17, 2010 Packet Loss <= 0.1% Hi Bo *POP* How *POP*e you? Hi Bo How you?

SHARKFEST ‘10 | Stanford University | June 14–17, 2010 Jitter <= 20ms Better late than never? No. May as well be lost.

SHARKFEST ‘10 | Stanford University | June 14–17, 2010 Demos

SHARKFEST ‘10 | Stanford University | June 14–17, 2010 Thanks! This presentation will be downloadable from the Sharkfest website.