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, 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. :)

FreeSWITCH, Jitsi, ZRTP

FreeSWITCH is a component of BigBlueButton that routes SIP calls. 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