[asterisk-users] C450 broken rtp handling

[Stanisław Pitucha]

Hello,
I've got a problem with rtp handling by siemens c450 and similar. I experience a couple seconds of silence between early media and normal call (normal call's rtp is dropped by phone). This is caused by SSRC changing (even though marker bit is set). I have all relevant patches applied - it still happens on 1.4.21.1 and every version before that. Especially http://bugs.digium.com/view.php?id=12570 doesn't change anything, because call is p2p bridged.

Issue can be fixed by forcing use of the same ssrc in ast_raw_write and bridged rtp writes and my custom patch works, but I don't want to use it if it can be done in some other way. Is there a way to force treating outgoing rtp as one stream, instead of switching source after early media? Is there a way to do it without resigning from p2p bridging?

Thanks for ideas,
Stan