1. Proving SIP Interoperability Networld+Interop Las Vegas 2004 iLabs Team

2. 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

3. 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.

4. Add Several SIP Proxy Servers DNS DHCP TFTP We installed a total of 6 SIP proxy servers, although we did not complete interoperability testing with 3Com Asterisk (Digium) SIP Express Router (iptel.org) Nortel Avaya Cisco 3Com

5. Finally, add SIP phones (and gateways) DNS DHCP TFTP Asterisk SER Nortel Avaya Cisco 3Com Nortel 3Com Snom Pingtel Siemens Cisco Polycom Pulver Avaya ipDialog Dlink MultiTech Grandstream

6. Start Testing Within Each SIP Proxy DNS DHCP TFTP 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

7. 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

8. We tested all types of phones Including hard phones, soft phones (software on laptops) and FXS gateways

9. Once we saw good interoperability…  We started to add other links, including T1 connections between devices (this took a bit of debugging) …

10. 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

11. 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 SER D-Link Cisco

12. You can connect to the “real world” as well TellMe Voice Information Service Internet PSTN Interop Link to PSTN Dial 9-1-800-555-8355 to connect to TellMe over the PSTN Dial 8-411 to connect to TellMe over the Internet Dial 9 plus your own home phone number to call out over the PSTN

13. We have prepared five white papers to help you  What is SIP?  What is ENUM?  Getting Started with SIP  SIP Migration Strategies  SIP Resources

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