I have a trouble with srtp:
if I start a call with srtp all is ok,
but if another user initiates call :
I answer - all ok (key in OK ends with "1S3" ),
I press HOLD - Jitsi generates new key for INVITE(ends with "D7K").
when I press unHOLD - Jitsi sends a second INVITE (ends with "D7K").
but on the other side can not hear me, because Jitsi sends srtp in key
ends with "1S3" )
If I initiate the call - Jitsi generate 1 key for a INVITE and uses it for
reIINVITEs and all is ok.
Hmm. AFAIK it is perfectly legal to change the encryption key during a
reinvite as this is basically a complete replacement of the media streams.
(Emil, can you confirm this?)
I remember having tested hold with multiple clients and hardphones. None of
them had a problem with this. However, they were either connected directly
(registrarless SIP) or with Kamailio as a SIP router. I repeated a test with
MicroSIP just now and it worked fine.
But I could reproduce your scenario when Asterisk was in the middle. Which
leads me to believe that it messes up with the reinvites and doesn't use the
new keys. Can you either confirm you're too using Asterisk or describe your
client setup in more detail?