[asterisk-users] changing of ssrc between early-media and call media

[Francesco Castellano]

Greetings,

upgrading from 1.4.17 to 1.4.19 some asterisk gateway of ours (used
for gatewaying ISDN-PRI and SIP), I noticed an annoying thing: when
the PSTN party answers, for a few seconds (4/5 sec typical) some SIP
client could not hear anything (the ringing was heard well!), then the
audio comes back again with no problem.

Looking for any differences between the behaviour of version 1.4.17
and 1.4.19 I found that in the new version the RTP stream changes SSRC
between the early media session and the actual call session. This
seems to me quite pretty, and a major part of SIP clients seems not to
be disturbed by it. Anyway I'd like to ask you a couple of things on
this issue:

1) Is the changing of ssrc standard compliant? (I suppose yes, because
the source changes from the Asterisk generating the ringing tone to
the remote PSTN party actual speech, but I am not sure at all on
this).

2) Do you know a way for avoiding such a change, in the meanwhile the
SIP clients having problems will be appropriately patched? Maybe, I
don't know, suggesting the PSTN to generate the ringing tone: how?

Thanks,
Francesco Castellano