Wanna know SIP? Read the RFC!
When I first configured Asterisk a few years back, it took me a lot of time to set it up properly because of many technical issues I experienced: echo, jitter, authentication issues, caller id problems, IAX vs. SIP (Pick SIP), NAT, dial plan and what not.
From Wikipedia: In computer network engineering, a Request for Comments (RFC) is a memorandum published by the Internet Engineering Task Force (IETF) describing methods, behaviors, research, or innovations applicable to the working of the Internet and Internet-connected systems.
Many new comers to VoIP who decide to mess with Asterisk usually deal with the same problems I dealt with, and usually making a basic interconnection comes with some pain and quite a bit of debugging.
So sure, you have www.voip-info.org which is an excellent reference on VoIP, and you have plenty of forums and nowadays you even have an official Digium Asterisk Documentation.
All of that is excellent, but the one thing I regret not doing in the first place, prior to messing with, reading about and beating my head off over the asterisk conf files is the SIP RFC.
RFC 3261 (“SIP: Session Initiation Protocol”) is the bible of SIP, the most popular signalling protocol for VoIP today. If you don’t like the text version, you can find HTML and PDF versions as well. While RFC are usually highly technical documents created by engineers for engineers, this particular RFC comes with a very nice introduction on how SIP works. Once you become familiar with the terminology of the RFC, I promise you, things will become much simpler.
With that, you can find other useful references about VoIP and networking here below:
- Digium Asterisk Documentation
- Building Telephony Systems with OpenSIPS 1.6
- FreeSWITCH 1.0.6 – Build Robust High Performance Telephony Systems
- FreePBX 2.5 Powerful Telephony Solutions
- How to Use Wireshark to Capture, Filter and Inspect Packets
- Cisco Networking Simplified (2nd Edition)
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