[jitsi-dev] Google Talk call issues in latest snapshots (Debian Linux 32-bit)


With this latest snapshot(s) seems that the Google Talk calls have some issues - namely only one side (me) can hear the other. Now this behavior is old in my case, but it usually lasts for a few seconds, then the SRTP key exchange is succesful and all works.

Today i had to try 4 times to call the other side in order to have a succesful conversation (every time i called and i heard the other from the second she picked up, but she couldnt hear me). This lasted 15-20 seconds before we gave up and tried again.

Here is the in-call info from the succesful conversation:

Call information : Identity : me@gmail.com (Google Talk)
Signalling call transport : TLS
other@gmail.com/jitsi-3ai63C668925 :
Call duration : 00:02:53
Audio info : Media stream transport protocol : UDP / SRTP (Key exchange protocol: ZRTP AES-CM-256/DH3K)
Codec / Frequency : opus / 48000 Hz
ICE candidate extended type : stun server reflexive Local host IP / Port: :
Local reflexive IP / Port: : MY.REAL.IP.ADDR/5008
Remote host IP / Port: :
Bandwith : ? 54 Kbps ? 55 Kbps Loss rate : ? 1% ? 0%
RTT : 381 ms
Jitter : ? 19 ms ? 20 ms
ICE Processing State: : Completed
Total harvesting time: : 387 ms Harvesting time GoogleTurnCandidateHarvester: : 64 ms Harvesting time GoogleTurnSSLCandidateHarvester: : 374 ms Harvesting time HostCandidateHarvester: : 2 ms Harvesting time JingleNodesHarvester: : 6 ms Harvesting time StunCandidateHarvester: : 56 ms Harvesting time TurnCandidateHarvester: : 64 ms Harvesting time UPNPHarvester: : 1506 ms

See that both computers have the same local IP class - they both are in the same VPN. I assume in this case it should be simple to plot the calls path, but for some reason sometimes the call wants to go via the internet on one side and via vpn the other side - maybe thats why it failed (unfortunately i lost the unsuccesful call info, forgot to paste it).
Is there some issue in the pathfinding logic if multiple routes are available (internet vs vpn, maybe in some instances the direct route is faster than the vpn)? Skype channeled the calls via the vpn every time.

PS1. Why is that i can hear the other every time (regardless who calls who) even before the srtp key exchange happens, but they couldnt hear me until the exchange is completed?

PS2. Please add line endings to the info panels text, it is a mess when copy-pasted.


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