On Wed, May 26, 2021 at 08:26:04PM -0500, Keith wrote:
That is - that media endpoints in INVITEs/200s and
re-INVITEs from a SIP
UA to osmo-sip-connector just ended up being transparently communicated
to osmo-bts. This made it really easy to take the pbx out of the audio
stream.
yes, but it unfortunately also measn you cannot have separate, non routed
networks between RAN and core network, which anyone with a reasonable interest
in network security always wants to have, so you have to put some kind of
RTP proxy like osmo-mgw co-located at the BSC level, as the BSC is the only
element that has connectivity with the CN.
freeswitch even has a console command for this:
"uuid_media" can switch
the pbx in and out of the stream during the call. [...]
interesting to know, thanks.
TS 29.164 (section 6) says that the GCR should be
encapsulated in a
binary encoded ISUP payload, there are pointers in the spec to ITU
Q.1912.5 (Section5 and thereabouts) So the ISUP messages should be added
to SIP messages, requiring a multipart MIME attachment to the INVITE
with the SDP and the ISUP. Also see RFC3204
oh my. Seriously? I just re-read the specs starting from 23.284 for
LCLS and ideed you seem to be correct :(
So I'm thinking that I need to serialise the GCR
and add it to an "X-"
SIP header, so we at get it back on the B-leg.
makes sense.
So yes, I need to just do this, but I got stalled a
bit again with my
lack on programming skills trying to learn how to serialise the GCR the
"right" way.
isn't there already some kind of function to print it to a string?
- but once I figure that out, then I can at least test
what
happens and see how the MSC and MGW behaves with the MNCC messages that
result from SIP re-INVITES.
Which re-invites? I think it might be useful to create a ladder diagram
of how you think this all works together.
--
- Harald Welte <laforge(a)osmocom.org>
http://laforge.gnumonks.org/
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