concepts
http://www.jdrosen.net/ietf.html
SER, OpenSER, OpenSIPs Kamailio
http://sip-router.org/history/
http://by-miconda.blogspot.com/2010/01/best-of-new-in-kamailio-300-12.html
Kamailio handles only the SIP signaling traffic, no media (e.g., audio or video) processing, therefore using Kamailio for telephony is just a subset of what can do. link
media proxy vs rtp proxy
http://www.mail-archive.com/users@lists.kamailio.org/msg00821.html
http://www.voipuser.org/forum_topic_4994.html
freeSWITCH vs yate
http://telecommusings.blogspot.com/2009/10/voip-why-you-should-try-freeswitch.html
code and bandwidth
config phones
SIP websockets WebRTC
http://sip-on-the-web.aliax.net/
- jsSIP sip over websockets
- WebRTC for RTP
- phono.com xmpp javascript stack
http://www.webrtc.org/reference/architecture
security
http://www.infosecwriters.com/text_resources/pdf/Zfone_SSotillo.pdf
billing attach in sip
http://www.csc.ncsu.edu/faculty/jiang/pubs/WOOT07.pdf&pli=1
http://www.cs.columbia.edu/~dgen/papers/conferences/conference-10.pdf
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