Building Telephony Systems with OpenSER is a new book from Packt, which acts as a step-by-step guide to building a high performance Telephony System. Written by Flavio E. Goncalves, this book teaches users how to develop a fast and flexible SIP server using OpenSER, an open-source VoIP server based on the Session Initiation Protocol (SIP), an application-layer control (or signaling) protocol for creating, modifying, and terminating sessions with one or more participants, including internet telephone calls, multimedia distribution, and multimedia conferences. This book is a well illustrated, step-by-step guide to building a SIP based network using OpenSER.
Let me be the first one to say this:
FreeSWITCH ROCKS!
Much better than Asterisk, it’s SIP stack is 100% RFC compliant and complete… it does STUN, TCP/UDP, SRTP, etc, they use the Sofia SIP stack.
Nokia also uses Sofia SIP, another great thing about FreeSWITCH is that it uses PCRE for regular expressions, it has event sockets, it’s a soft-switch so it can scale from a PBX to a soft-switch, etc.
It also has YAML support for writing dialplans now, the rest is XML, etc.
I highly recommend FreeSWITCH to anyone that is interested in VoIP and telecommunications.
http://freeswitch.org/
FreeSWITCH vs Asterisk
http://freeswitch.org/node/117
It’s the Apache of VoIP =D
I also forgot to mention that FreeSWITCH does not require any Zaptel for conference.
Asterisk requires Zaptel for it’s timing source in order to do conferencing.
FreeSWITCH has it’s own internal API for timing, and it does soft-based conference just fine.
It’s the best application I ever used for VoIP.
Which is entirely irrelevant to a discussion of OpenSER.
Of course that it’s relevant.
No it isn’t. FreeSwitch is a soft switch, not a SIP router, and regardless, the book being reviewed is about OpenSER, not FreeSwitch. You guys spamming about Freeswitch on any and every article having something to do with VoIP all over the web gets old fast.