=Default configuration=
* [http://wiki.freeswitch.org/wiki/Default_config Default configuration]: includes picture for structure
=book=
在google group share下面有一本freeswitch cookbook,里面提及了很多基本概念不错，FreeSWITCH.Cookbook.pdf建议阅读：
* chapter1,有关call的一些基本概念。系统怎么处理的，怎么调用和加入一些基本的logic.
* chapter3: 有关一些call detail records,应该是做log方面可能需要。进一步研究。
* chapter4: 有关external control。需要阅读，讲了怎么建立 inbound/outbound event socket connection.
=Limit=
Freeswitch uses limit functionality to control concurrent calls on users and so on. Refer: <br/>
* [http://wiki.freeswitch.org/wiki/Limit limit]
=一些freeswitch应用里面出现的term=
* [http://wiki.freeswitch.org/wiki/Dialplan dialplan]: 简单来说就是不同的拨号规则
* [http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridge bridge]: Provides the ability to bridge two endpoints. Generally used to route an incoming call to one or more endpoints. 估计我们建立通话，和对通话做很多处理都需要用到。
*
=freeswitch event 相关=
* [http://wiki.freeswitch.org/wiki/Mod_event_zmq mod_event_zmq]：据说是另外一种类似inbound/outbound event socket.他利用了ØMQ library. 可以考虑使用以下。
* [http://wiki.freeswitch.org/wiki/Mod_xml_curl mod_xml_curl]: dynamically control the behavior offreeswitch
* [http://wiki.freeswitch.org/wiki/Mod_commands commands]: core commands for freeswitch cli extension.
* [http://wiki.freeswitch.org/wiki/Modules#Event_Handlers event handlers]: a list of modules for event handler.
* 主要很多event还需要参考对应的功能module.

=freeswitch 用户相关=
* [http://wiki.freeswitch.org/wiki/Mod_xml_curl mod_xml_curl]: 里面也包括了Authorization和ACL。
* [http://wiki.freeswitch.org/wiki/Mod_ldap mod_ldap]:据说是out of date,而且quality不够production standard.
* mod_xml_ldap: 有代码，但是没有更多的信息了，应该是dead了。
* [http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide display_directory]:好像直接和用户注册的信息有关，需要验证
* [http://wiki.freeswitch.org/wiki/External_profile external profile]:  The external profile handles external or outbound registrations, to a SIP provider. The SIP provider sends calls to you, and you send calls to your provider, through the external profile.  great starting place for learning about two very important variables that aid in NAT traversal.

=freeswitch audio conference相关=
* [http://wiki.freeswitch.org/wiki/Mod_conference mod_conference]:有关freeswitch的conference mode相关。我们需要做research，怎么样通过inbound/outbound socket来控制。
* [http://wiki.freeswitch.org/wiki/Sofia sofia]: think as sip.
=freeswitch multicast相关=
* 在freeswitch里面，multcast名称是multcast paging.
* [http://wiki.snom.com/Interoperability/PBX/FreeSwitch#Paging_.28Multicasting.29 softphone]应用multicast paging的一个例子
* [http://wiki.freeswitch.org/index.php?title=Special:PdfPrint&page=Variable_esf_multicast_ip esf_multicast_ip]：With mod_esf you are able to provide a multicast stream to your network from your Freeswitch
* [http://wiki.freeswitch.org/wiki/Multicast_Paging multicast pagging]: freeswitch里面一些multcast pagging叙述
* [http://wiki.snomone.com/index.php?title=Multicast_Paging multcast pagging defintion]:定义。

=freeswitch flash support=
* [http://wiki.freeswitch.org/wiki/Mod_rtmp Flash support]: 这个module适用于支持flash的。

=freeswitch other modules=
* [http://wiki.freeswitch.org/wiki/Modules all modules]: a list of modules of freeswitch.
* [http://wiki.freeswitch.org/wiki/Authoring_Freeswitch_Modules create a new module]: 可以新建一个我们自己的freeswitch module.

=freeswitch media flow=
* [http://wiki.freeswitch.org/wiki/Proxy_Media proxy media]:主要介绍了几种media flow control在freeswitch里面，也就是RTP携带的数据。
* [http://wiki.freeswitch.org/wiki/Bypass_media by pass media]: by pass mode指的什么。注意，对于bypass mode来说，很多操作都无法执行，比如录音等。
=freeswitch issues=
* [http://wiki.freeswitch.org/wiki/RTP_Issues RTP]:issue.

=NAT issue=
* [http://wiki.freeswitch.org/wiki/NAT_Traversal freeswitch NAT tranversal]
* [http://wiki.freeswitch.org/wiki/NAT some other topics on NAT]: freeswitch have UPnP and NAT-PMP or using STUN to detect external IP. Also it said that if client sip is behind NAT, freeswitch will send a alive request every 30s to keep it alive.

=message=
* [http://wiki.freeswitch.org/wiki/Mod_dingaling mod_dingaling]:xmpp registration that can be used for Google talk or any other XMPP integration.
=embed freeswitch=
=GUI=
[http://wiki.freeswitch.org/wiki/Freeswitch_Gui A list of GUI]

=codec相关=
* [http://wiki.freeswitch.org/wiki/Codecs codec wiki]
* [http://wiki.freeswitch.org/wiki/Codec_Negotiation codec negotiation]
* [http://wiki.freeswitch.org/wiki/VAD_and_CNG Voice Activity Detection & Comfort Noise]
* [http://en.wikipedia.org/wiki/RTP_audio_video_profile audio video profile]
* [http://www.faqs.org/rfcs/rfc5219.html mp3 on rtp rfc5219]
* [http://www.live555.com/rtp-mp3/ sample] of rfc 3119(previous than rfc5219)
* [http://ffmpeg.org/doxygen/trunk/rtpdec_8c-source.html rtp mp3 on ffmpeg]
* [http://tools.ietf.org/html/rfc5215 vorbis rtp]
==implementation==
* found ffmpeg have adu called : mp3adu decoder only.
* [http://www.live555.com/liveMedia/#config-unix live555] have a encoder, let's try to work them togather.

=performance=
* [http://www.bandcalc.com/ VOIP calculator]
* [http://wiki.freeswitch.org/wiki/Real-world_results/ Real World Freeswitch Performance]

=Other=
* Auto-Answer: in sip header field: Call-Info: <sip:127.0.0.1>;answer-after=0 .. In side freeswitch, use <action application="export" data="sip_auto_answer=true" /> to turn on

=freeswitch call flow/channel/session flow=
* [http://wiki.freeswitch.org/wiki/Life_Cycle_of_a_Call life cycle of a call]
* [http://lists.freeswitch.org/pipermail/freeswitch-users/2006-August/028095.html email] about the session vs channel.

= API =
* [http://docs.freeswitch.org/ api]
= cluster =
* [http://old.nabble.com/Clustering-FreeSWITCH-td20218422.html mail] about freeswitch cluster dicussion.
* [http://freeswitch-users.2379917.n2.nabble.com/mod-conference-with-cluster-ODBC-td5887418.html mail] about freeswitch conference cluster discussion
* [http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS deployment] of freeswitch cluster