Hi,
We're using sipXtapi to act as a kind of VoIP gateway on an embedded
system. The gateway connects an IP interface with an ATM bus, with legacy
phone applications running on the ATM side.
So far, the gateway initializes a single SipUserAgent using
sipxInitialize(), and sipXmediaLib has been modified to use flowgraphs
which read/write ATM sockets. Most things are working just fine, except for
two issues:
- by default, only one connection can be active, but as we're
acting as gateway, all audio is allowed to stream for all
connections (there is no speaker/mike connected to the gateway
itself)
- when dialling from one SIP phone connected to another SIP
phone on the same gateway, the INVITE is rejected, as sipXtapi
thinks it is an invite to itself (the IP's are the same, but
the bit before the @ isn't)
Issue #1 was easily solved by never "disabling" a flowgraph when the
upper-level API requested so. As we're using our own flowgraphs everything
seems to be working fine.
For #2 we can't seem to find an "easy" solution, other than running two
SipUserAgents on the gateway, one for outgoing and one for incoming calls.
This however might not feasible to performance constraints (it is something
like a 50Mhz processor...)
Is there someone on this list which has a better knowledge of sipXtapis
inner workings to suggest a work around for issue #2?
Thanks,
Herman
(PS: we're using a tagged version from a few months ago)
-----------------------------------------------------------------
Herman Kuiper - m: [EMAIL PROTECTED] - w: http://www.frontier.nl
Beech Ave 162 - 1119 PS Schiphol-Rijk - t/f: 020-6589034/6142816
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