BigBlueButton

BigBlueButton is a video conferencing server. It is built on a collection of FLOSS projects including FreeSWITCH, which also gives it the ability to route SIP calls and XMPP sessions, giving users the ability to do text chat, VoIP, and video chat with end to end encryption.

I am attempting to get all the pieces working on a personal server and on a machine at HSL. I have built up a fair bit of state, and I'm figuring to use this page for notes so I don't lose my place. Perhaps it will also act as breadcrumbs for future explorers if my attempt goes pear-shaped. :)

SIP

FreeSWITCH, Jitsi, ZRTP

FreeSWITCH is a SIP router that is bundled with BigBlueButton. I have it installed and routing calls between softphones, but I do not have everything working yet. I have not made a successful SIP call between two different clients, I do not have video working through FreeSWITCH (though direct dialing with Jitsi works), and I do not have ZRTP working through FreeSWITCH (again, working with direct Jitsi-to-Jitsi calls).

I have just grabbed the Jitsi source code, wiped my installs, and re-verified that direct Jitsi-to-Jitsi calls have encryption and video. Now I am going to document the failures and see if I can find any useful info in the logs. Jitsi logs are pretty sparse, even with debug enabled, so I may wind up adding log statements.

Jitsi-to-Jitsi, Direct Connect, ZRTP+Video

Call information :
 Identity : bob (RegistrarLess SIP)
 bob@192.168.1.10 :
 Call duration : 00:22:29
 Audio info :
 Media stream transport protocol : UDP / SRTP (Key exchange protocol: ZRTP AES-CM-128/DH3K)
 Codec / Frequency : opus / 48000 Hz
 Local IP / Port : 192.168.1.6 / 5000
 Remote IP / Port : 192.168.1.10 / 5000
 Bandwith : ↓ 6 Kbps ↑ 28 Kbps
 Loss rate : ↓ 0% ↑ 0%
 Packets decoded with FEC : 1
 Packets currently being discarded : 0%
 Number of discarded packets : 2 (0 late, 1 full, 0 shrink, 1 reset)
 Adaptive jitter buffer : enabled
 Jitter buffer : ~40ms;
 currently in queue: 1/4 packets
 RTT : 7 ms
 Jitter : ↓ 2 ms ↑ 0 ms
 Video info :
 Media stream transport protocol : UDP / SRTP (Key exchange protocol: ZRTP AES-CM-128)
 Video size : ↓ 640 x 480 ↑ 640 x 480
 Codec / Frequency : H264 / 90000 Hz
 Local IP / Port : 192.168.1.6 / 5002
 Remote IP / Port : 192.168.1.10 / 5002
 Bandwith : ↓ 35 Kbps ↑ 46 Kbps
 Loss rate : ↓ 0% ↑ 0%
 Packets decoded with FEC : 0
 Packets currently being discarded : 0%
 Number of discarded packets : 0 (0 late, 0 full, 0 shrink, 0 reset)
 Adaptive jitter buffer : enabled
 Jitter buffer : ~100ms;
 currently in queue: 0/11 packets
 RTT : 396 ms
 Jitter : ↓ 2 ms ↑ 3 ms

Jitsi-to-Jitsi, Via FreeSWITCH, fail ZRTP and Video

Call information :
 Identity : 1003@192.168.1.7 (SIP) Signalling
 call transport : UDP
 
 1004@192.168.1.7 :
 Call duration : 00:01:08

 Audio info :
 Media stream transport protocol : UDP / RTP
 Codec / Frequency : G722 / 16000 Hz
 Local IP / Port : 192.168.1.6 / 5028
 Remote IP / Port : 192.168.1.7 / 31332
 Bandwith : ↓ 68 Kbps ↑ 64 Kbps
 Loss rate : ↓ 0% ↑ 0%
 Packets decoded with FEC : 0
 Packets currently being discarded : 0%
 Number of discarded packets : 7 (0 late, 7 full, 0 shrink, 0 reset)
 Adaptive jitter buffer : enabled
 Jitter buffer : ~40ms;
 currently in queue: 0/4 packets
 Jitter : ↓ 1 ms ↑ 0 ms 

Possible Solution: FreeSWITCH: Proxy Media

http://wiki.freeswitch.org/wiki/Proxy_Media

Looking like a promising path. Proxy Media basically means, "Hey, FreeSWITCH: You're just a dumb router, not a trusted participant in my telephone call."

I now have made video calls from Linphone/Linux to Linphone/Cyanogenmod, and have some promising looking ZRTP (encryption) error messages connecting from Jitsi/Linux to Jitsi/OS-X.

Probably no more progress for a few days, business trip.

ZRTP Support Notes

Linphone

https://github.com/bitzeppelin/linphone-sdk/blob/master/README.zrtp

Readme on the inclusion of GNU ZRTP in the Linphone codebase. Includes notes on enabling ZRTP at compile time. Perhaps published binaries do not include ZRTP support?

XMPP

XMPP is the Jabber protocol, initially created for text chat. It has been extended to include voice and video with Jingle.

Text

XMPP was originally designed for text.

Jingle

Jingle is an extension to XMPP that enables voice, video, and other new features.

ICE

ICE is a method for negotiating NAT traversal (getting through firewalls) for clients that are using multiple channels.

http://xmpp.org/extensions/xep-0176.html

Encryption

ZRTP can be used with XMPP, and I think it has other options for end-to-end encryption.