1、doubango官网：http://www.doubango.org/

2、doubango是一个开源的VOIP基础平台， 并能用于嵌入式和桌面系统的开源框架，该框架使用ANSCI-C编写，具有很好的可移植性。

3、根据doubango的框架、特点与linphone的对比，显然doubango更适合做视频会议系统和视频会议终端的选择，理由？根据linphone的rtp会话管理可见只适合2方进行视频通话，如要支持多方（>3）视频会议，需MCU配合server进行音视频转发和mix，linphone对于nat穿透的方案只有stun和转发，linphone适合更适合当做一款网络电话终端供普通网民当做视频电话用。

4、doubango包含了 SIP/IMS (VoIP) 的客户端、服务器端的组件。

 

Client-side components
sipML5	HTML5 SIP client using webrtc2sip Gateway. Enjoy our live demo ?
clik2dial	A complete Click-to-Call Solution using webrtc2sip Gateway and sipML5. Enjoy our live demo ?
webrtc4all	WebRTC extension for Safari, Opera, Firefox and IE.
Boghe	SIP video client for Windows Phone 8 and Surface Pro IMS/RCS Client for Windows XP, Vista, 7 and 8
iDoubs	SIP/IMS VideoPhone for iOS (iPhone, iPad and iPod Touch) and MAC OS X.
IMSDroid	SIP/IMS Client for Android.
libSigComp	Signaling Compression (SigComp) library
Server-side components
webrtc2sip	Smart SIP and Media Gateway to connect WebRTC endpoints to any SIP-legacy network. Enjoy our live demo ?
telepresence	Open Source TelePresence system with a smart and powerful MCU.  Enjoy our live demo ?
Flash2IMS	Adobe? Flash? to SIP/IMS Gateway.

 

5、doubango的主要特点

? Powerful MCU (Multipoint Control Unit) for audio and video mixing

? Stereoscopic (spatial) 3D and stereophonic audio
? Full (1080p) and Ultra (2160p) HD video up to 120fps
? Conference recording to a file (containers: *.mp4, *.avi, *.mkv or *.webm)
? Smart adaptive audio and video bandwidth management
? Congestion control mechanism
? SIP registrar
? 4 SIP transports (WebSocket, TCP, TLS and UDP)
? SA (direct connection to SIP clients) and AS (behind a server, such as Asterisk, reSIProcate,
openSIPS, Kamailio…) modes
? Support for any WebRTC-capable browser (WebRTC demo client at http://conf-call.org/)
? Mixing different audio and video codecs on a single bridge (h264, vp8, h263, mp4v-es,
theora, opus, g711, speex, g722, gsm, g729, amr, ilbc)
? Protecting a bridge with PIN code

? Unlimited number of bridges and participants

? Connecting any SIP endpoint
? Easy interconnection with PSTN
? NAT traversal (Symmetric RTP, RTCP-MUX, ICE, STUN and TURN)
? RTCP Feedbacks (NACK, PLI, FIR, TMMBN, REMB…) for better video experience
? Secure signalling (WSS, TLS) and media (SDES-SRTP and DTLS-SRTP)
? Continuous presence
? Smart algorithm to detect speakers and listeners
? Different video patterns/layouts
? Multiple operating systems (Linux, OS X, Windows…)
? 100% open source and free (no locked features)
? Full documentation
? …and many others

    6、doubango架构图

说明：

tinySAK (Swiss Army Knife): Utilities functions (SHA-1, MD5, HMAC, String, List, Timers, Thread, Mutex, Semaphore, ...)
tinyNET: Networking (DNS, DHCPv4/v6, STUN, TURN, ICE, ENUM, Sockets, ...)
tinyHTTP: HTTP stack (CONNECT, DELETE, GET, HEAD, OPTIONS, PATCH, POST, PUT, TRACE, ...)
tinyXCAP: XCAP stack (AUID manager, URL generator) without XML parser (See Java code for parsers)
tinyIPSec: IPSec SA manager. Useless for Android but you MUST have it
tinySMS: SMS over IP (SM-TL, SM-RL) for IMS/LTE networks
tinySIGCOMP: Signaling Compression
tinySDP: SDP protocol
tinyRTP: RTP/RTCP protocols
tinyMSRP: MSRP protocol (Chat and File Transfer)
tinyMEDIA: Media plugins manager (Audio, video, Codecs, sessions, MSRP, QoS, ...)
tinyDAV(Doubango Audio Video): Media plugins implementation
tinySIP: SIP/IMS stack