AU-9 (Walberg) Expose VoIP Problems with Wireshark

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Transcript AU-9 (Walberg) Expose VoIP Problems with Wireshark

Expose VoIP Problems With Wireshark

June 18, 2009

Sean Walberg

Network Guy | Canwest

SHARK

FEST

'09

Stanford University June 15-18, 2009

SHARKFEST '09 | Stanford University | June 15 –18, 2009

Without tools, VoIP is a black box

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Wireshark lets you peek inside

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VoIP is just another application

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(but it has special requirements)

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About Me

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About You

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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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Local Loop The old way

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The old way Off Hook Dialtone

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The old way Dialing Digits

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The old way RING – 90v@20Hz

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The old way

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The VoIP way

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The VoIP way

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The VoIP way

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The VoIP way ZZZZZZ

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So there are two parts to VoIP • • Signaling – SIP – H.323

– MGCP – SCCP – Proprietary Voice (Bearer) – RTP (G.711, G.722, G.729a,…)

SHARKFEST '09 | Stanford University | June 15 –18, 2009

Jitter, Delay, and Loss, oh my!

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Loss

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Delay Never underestimate the bandwidth of a station wagon loaded with backup tapes.

(the delay is a different matter)

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Jitter

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Delay Jitter != Delay Jitter

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About VoIP Capturing VoIP Signaling RTP

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Location, Location, Location

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Just a simple network

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The signaling traffic takes a different path from the RTP traffic

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Or, it might do this

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Same conversation, different perspectives Here you see inbound latency and jitter, but nothing on the outbound Here you see inbound latency and jitter, but nothing on the outbound

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NAT changes the address Src=C Dst=D Src=A Dst=B The address changes within the cloud!

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Set your capture filters

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The Packet List window

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Summaries are displayed here

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By the way… If the signaling or the voice is encrypted, you won’t be able to decode it.

Sorry.

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Quality of Service for VoIP networks

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Use color to show QoS problems

View -> Coloring Rules SHARKFEST '09 | Stanford University | June 15 –18, 2009

Add a column for DSCP Signaling Tagged RTP Untagged RTP

Edit -> Preferences User Interface->Columns SHARKFEST '09 | Stanford University | June 15 –18, 2009

Are you running a proprietary PBX?

Edit -> Properties, Protocols -> RTP SHARKFEST '09 | Stanford University | June 15 –18, 2009

Use the Packet Details pane to see what’s inside the packet

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About VoIP Capturing VoIP Signaling RTP

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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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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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Demos

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About VoIP Capturing VoIP Signaling RTP

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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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DTMF • • Compressing DTMF is bad So many different ways to carry the digits out of band, look for them in traces

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Three factors that affect voice quality

Latency <= 150ms (one way) Jitter <= 20ms Packet loss <= 0.1% SHARKFEST '09 | Stanford University | June 15 –18, 2009

Latency <= 150ms (one way) Jitter buffer, Transcoding delay Path delay Serialization delay Hi, how are you?

Hello? Oops, sorry, go ahead Fine, I oh hello, go ahead

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Packet Loss <= 0.1%

Hi Bo *POP* How *POP*e you?

Hi Bo How you?

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Jitter <= 20ms Better late than never? No. May as well be lost.

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Demos

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Thanks!

[email protected]

@seanwalberg This presentation will be downloadable from http://lovemytool.com and http://cacetech.com

SHARKFEST '09 | Stanford University | June 15 –18, 2009