(hopefully) fixed a sporadic crash in sdp_message_parse()
- RTP timeouts on some PBX systems that include an video
stream, but the called UA has no video capability.
behind the same siproxd to have conversation together
UA1 -->--\ /-->--\
siproxd Registrar
UA2 --<--/ \--<--/
- Redone code for evaluation if a received packet
if coming from inbound or outbound interface
- RTP stream are now identified by call_id AND
USERNAME of the contact header. This provides
support for RTP proxying between 2 UAs sitting on the
inbound network. -> Calls between local UAs going via
siproxd should now work.
UA1 -->--\
siproxd
UA2 --<--/
- Rewriting of SUBSCRIBE should now work.
- Removed obsolete prototypes from rtpproxy.h
- If the RTP stream in one direction is found to be
stopped (sendto()) also stop the opposite direction
- RTPPROXY correction: match RTP ports crosswise -
use one single port (and socket) on each side (inbound/
outbound) to send and receive RTP traffic for every
active stream (patch by Christof Meerwald).
a completely statically linked executable
- REGISTER takes honors the expires parameter
of the contact header
- Contact header of REGISTER response must be
rewritten back to the local (true) URL
his work on this). Up to now, only the RTP *Relay*
has been tested (works with KPhone, BudgeTone)
- fix: SIP phones that allocate a random port for
incomming SIP traffic should now work (like BudgeTone)
- fix: some SIP phones do change the RTP port number
during a session (like KPhone during HOLD/unHOLD)
* have siproxd compile on Solaris and BSD/OS (more to come)
* ./configure option --with-libosip-prefix
* properly handle getopt_long()/getopt()
- First attempt of iptables support. Routines to add/remove
DNAT entries must still be done.
