Proving SIP Interoperability

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Proving SIP Interoperability

• NetworldInterop
• Las Vegas 2004
• iLabs Team

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To Test VoIP,Start with IP
We built a very standard looking IP network using
off-the-shelf hardware from Extreme, Check Point,
3COM, and Cisco We added wireless from Aruba and
KVM services from Avocent
We also used Unix as a platform for
infrastructure services, such as DNS, DHCP, and
TFTP
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Add SeveralSIP Proxy Servers
SIP proxy servers are important for call setup
during SIP calls. These servers often have other
features built into them, such as voice mail
systems or IVR (interactive voice response) tools.
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Add SeveralSIP Proxy Servers
Asterisk(Digium)
SIP Express Router (iptel.org)
We installed a total of 6 SIP proxy servers,
although we did not complete interoperability
testing with 3Com
Nortel
Avaya
Cisco
3Com
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Finally, add SIP phones(and gateways)
Nortel 3Com Snom Pingtel Siemens Cisco Polycom Pul
ver Avaya ipDialog Dlink MultiTech Grandstream
Asterisk
SER
Nortel
Avaya
3Com
Cisco
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Start Testing Within Each SIP Proxy
SIP Domain A
SIP Domain B
We called from each SIP phone to every other
phone. Because voice uses a different protocol
than signalling, we also verified that voice
works in each direction
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Then, Call Between Each SIP Proxy
SIP Domain A
SIP Domain B
Calling between each SIP domain shows
interoperability both of phones and of SIP proxy
servers
We achieved near-100 interoperability when
testing calling and voice traffic between devices
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We tested all types of phones
Including hard phones, soft phones (software on
laptops) and FXS gateways
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Once we saw goodinteroperability

• We started to add other links, including T1
  connections between devices (this took a bit of
  debugging)

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Once we saw goodinteroperability

• We started to add other links, including T1
  connections between devices (this took a bit of
  debugging)

and a connection to the public switched
telephone network (PSTN) through the eNet
provider, Avaya
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Test interoperabilityfor yourself!
Pick up any phone and dial the four digit number
of any other
When you cross the red line, youre proving
interoperability of four vendors two SIP phones
and two SIP proxies
Cisco
D-Link
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You can connect to the real world as well
Dial 9-1-800-555-8355 to connect to TellMe over
the PSTN
Internet
Dial 9 plus your own home phone number to call
out over the PSTN
TellMeVoice InformationService
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We have prepared five white papers to help you

• What is SIP?
• What is ENUM?
• Getting Started with SIP
• SIP Migration Strategies
• SIP Resources

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Thanks for Visiting iLabsVoice over IP using SIP
Dustin Goodwin, Craig Watkins, Joel Snyder, Jan
Trumbo, Doug Moeller, and John Balogh