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Learning SIP Online
All the same name, but different. There is SIP to the IP phone, SIP to the PSTN, SIP for IP trunking between call managers and SIP for multimedia.Some of you have read Alan Percy's blogs, "SIP Interoperability: Why Is It So Hard To Achieve (Parts I, II and III) and discovered that SIP is not a single target, but a matter of target interpretation. The SIP standard, RFC 3261, is so broad and with so many options, learning SIP fundamentals should come first, then learning of the particular SIP interpretation you will acquire with the SIP vendor's product or service. There is much information, many white papers, documents and blogs covering SIP. There are textbooks and vendor booklets all covering some aspect of SIP.
I was looking for a vendor neutral presentation of SIP basics through troubleshooting. Graham Francis, CEO of Vocale Ltd and The SIP School, Birmingham, U.K. made their online curriculum available for my review. You can view the nine course descriptions and six demos here. The nine courses are:
1. Core SIP (this is for the beginner) 2. SIP-T and the PSTN* 3. SIP and VoIP 4. Firewalls/NAT and SBCs 5. SIP Security 6. SIP Trunking* 7. ENUM and DNS 8. Testing and Troubleshooting SIP 9. SIP and UC * These two courses focus on SIP trunking for PSTN/ITSP connections that has become very popular.
Courses 2, 3, 4 and 5 are for intermediate students. Courses 6, 7, 8 and 9 are the most advanced and assume considerable SIP knowledge. The on-screen animated material is accompanied with audio commentary. Each course allows the student to skip around the slides and select what is of most value without having go to through previous slides. The SIP School also offers a certification, the SIP School Certified Associate (SSCA). The SSCA exam consists of 55 questions. The certification is valid for 24 months.
The first course I accessed was "Firewall/NAT and SBCs." Most enterprises think of the firewall for connecting to the public Internet. I have also found that regulations for some industries require firewalls to be implemented between business units within an enterprise such an electrical utility with nuclear power generating plants. Companies that have contracts with the DHS or DoD also need internal firewalls among business units.
This course covers the problems of traversing these boundaries. A number of solutions are discussed including:
* STUN--Simple Transversal of UDP * TURN--Transversal Using Relay NAT * ICE--Interactive Connectivity Establishment * UPnP--Universal Plug and Play * Media Proxy * Application Level Gateway * SIP Aware Firewall * SBC--Session Border Controller
Reviewing all these potential security solutions will help the enterprise to locate the appropriate solution.
This course does a very good job explaining not only the SIP issues, but also the Session Description Protocol (SDP) problems associated with RTP voice packet transversal with dynamic port number assignment. The course of 49 primary slides takes only 25 minutes to play without a pause. The average student can expect to devote at least one to two hours to cover all the material.
The second course I accessed was the "Testing and Troubleshooting SIP" course that I found useful and provided some new information for my own testing and troubleshooting problems. This course contains 58 primary slides with 5 to 10 secondary slides per primary slide that can be read if interested in further detail.
This second course covered installing and testing hard phones, softphones and live connections to an ITSP. The use of Wireshark is well covered for the testing and packet capturing of SIP VoIP calls. The course also goes into detail of the common messages associated with SIP interface problems. The main limitation is that only European ITSPs are accessed in the exercise for ITSP connection. Although the course takes only 30 minutes to play without a pause, the average student can expect to devote at least one to two hours to cover all the material.
Overall I liked the entire course series. As with a moving target such as SIP, the material will continually be enhanced. The one area I would like covered is SIP interoperability between call managers of the same and different vendors.
I am encountering clients that want to connect Avaya to Cisco call managers. The problems that will occur for this interconnection expands SIP testing into feature transparency such as the message waiting indicator, conferencing, call transfer as well as many other features across the call manager-to-call manager SIP trunks. I don't see the SIP school solving this issue, but I would like to see a course on what the interoperability problems would be. I also want suggestions of how to test SIP trunking between VoIP/IPT vendors call managers.