I have been playing around with FreeSWITCH for a long while but recently I gave up playing and implemented the system for a small research team, that other open source VoIP Systems could not easily serve. As well as for the research team that is spread over a few countries and time zones and me personally the mod_skypiax gave the final push to put the system into production. We actually started working with the sources for this project much earlier.
The team at freeswitch.org has officially released the FreeSwitch 1.0.4. With this release FreeSWITCH brings many improvements in stability and security and some additions like the mod_skypiax.
Here are some of the functions and additions that I found interesting;
OPAL is one of my favorite projects and have been a follower for a while. Now FreeSWITCH works together with Project Opal to create mod_opal, which adds H.323 functionality to FreeSWITCH as well as IAX2 support.
My favorite, mod_skypiax brings capability to send and receive Skype calls from FreeSWITCH to Skype network. The Skype module (mod_skypiax) uses the native Skype client for the target operating system. Currently, Windows and Linux are supported. FreeSWITCH's ability to transcode and resample other high quality codecs allows for high quality Skype calls coming over SILK codec to connect to SIP endpoints, be they free SIP clients or Commercial SIP Phones.
This capability will certainly promote the capability FreeSWITCH's capability as a telephony switch.
ZRTP will allow to encrypt calls between endpoints or systems. The ZRTP uses SRTP to accomplish the purpose and is co developed with Philip Zimmermann of PGP fame, which I have been using for ages. I have not tested this feature yet but looking forward to do so soon.
MRCP, or Media Resource Control Protocol, is provided with this module. This module utilizes the well-written UniMRCP library, authored by Arsen Chaloyen. The module is compliant with MRCP version 2 (SIP). MRCP allows for media servers to be separated from FreeSWITCH. Good module for tasks such as IVR or Text to Speech etc.
NAT Traversal Made Easy
NAT-traversal is important for SIP calls and is made easier with the automatic NAT handling feature included with the FreeSWITCH. Routers, switches and firewalls that support either UPnP or NAT-PMP can be polled by devices inside the network, allowing them to determine the external IP address. You can give up fighting with STUN now!
FreeSWITCH now supports real-time pre-paid billing applications with mod_nibblebill. The name of the module is telling: it monitors calls in progress and "nibbles" away at the available credit of a pre-paid user's account.
In addition to above, the FreeSWITCH has following features;
- Runs on Win32/MAC/UNIX
- IVR Application API
- 8kHz/16kHz/32kHz/48kHz Audio
- Soft Conferencing
- SIP B2UA/SRTP/TLS
- SIP BLF/SLA/PBX Features
- Async Audio
- Event/Logger Engine
- Real Time
- zRTP (libzrtp)