Transcript Proving SIP Interoperability

Proving SIP
Interoperability
Networld+Interop
Las Vegas 2004
iLabs Team
To Test VoIP,
Start with IP
DNS
DHCP
TFTP
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
Add Several
SIP Proxy Servers
DNS
DHCP
TFTP
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.
Add Several
SIP Proxy Servers
DNS
Asterisk
(Digium)
Nortel
Cisco
DHCP
TFTP
We installed a
total of 6 SIP
proxy servers,
although we did
not complete
interoperability
testing with 3Com
SIP
Express
Router
(iptel.org)
Avaya
3Com
Finally, add SIP phones
(and gateways)
DNS
Asterisk
Nortel
Cisco
DHCP
Nortel
3Com
Snom
Pingtel
Siemens
Cisco
Polycom
Pulver
Avaya
ipDialog
Dlink
MultiTech
Grandstream
TFTP
SER
Avaya
3Com
Start Testing Within
Each SIP Proxy
DNS
DHCP
TFTP
SIP Domain A
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
SIP Domain B
Then, Call Between
Each SIP Proxy
DNS
DHCP
TFTP
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
We tested all types of phones
Including hard phones, soft phones (software on laptops) and FXS gateways
Once we saw good
interoperability…
 We started to
add other links,
including T1
connections
between
devices (this
took a bit of
debugging) …
Once we saw good
interoperability…
 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
Test interoperability
for yourself!
Pick up any phone
and dial the four digit
number of any other
When you cross
the “red line,”
you’re proving
interoperability of
four vendors: two
SIP phones and
two SIP proxies
Avaya
D-Link
SER
Cisco
You can connect to the
“real world” as well
Dial 9-1-800-555-8355
to connect to TellMe
over the PSTN
Interop
Link to PSTN
Internet
Dial 9 plus your own
home phone number to
call out over the PSTN
PSTN
Dial 8-411 to connect
to TellMe over the
Internet
TellMe
Voice Information
Service
We have prepared five
white papers to help you
 What is SIP?
 What is ENUM?
 Getting Started with SIP
 SIP Migration Strategies
 SIP Resources
Thanks for Visiting iLabs
Voice over IP using SIP
Dustin Goodwin, Craig Watkins, Joel Snyder,
Jan Trumbo, Doug Moeller, and John Balogh