On 22/07/2023 05:37, Sipos Csaba wrote:
Hi Keith,
Hi.
(BTW, It doesn't help others to follow the thread when one replies
without context and with a subject line of "OpenBSC Digest...")
My question was targeting the Osmocom stack and not the (external)
PBX.
OK, Well there might be an issue in the question, "what codecs is
osmo-sip-connecter offering to the PBX when I set "codec-list fr3 fr2
fr1"? Actually I'd need to check that, especially with some patches that
are now in master but not released yet. With your setup, whatever
versions it might be using, you can very simply verify by taking a look
at your SIP, as I said before.
But given that you basically said it works with FR but not with fr3 in
there, one can only assume you are offering AMR and the PBX is not
accepting. So well, you need to target the PBX.
The second half of my question was somewhat vaguely tries to ask if it
is possible to do AMR when the call is MO-MT directly,
Yes, of course, that will work if you take the PBX/external MNCC out of
the loop, so having it work with the PBX in there requires that the PBX
accepts AMR for pass-through. Many years have passed since I used
asterisk, I don't remember how to configure it to do that, but it could
well be possible. Maybe even without having asterisk look at the rtp
stream at all. In times gone by FreeSwitch used to ship (maybe still
does) with a default pass-through only AMR codec implementation.
so in a nutshell: maybe there is a solution to
do GSM FR when the call is outbound from the mobile network and AMR
when it is mobile to mobile.
Yes, these are actually questions that can get somewhat complex. To be
honest, my level of clarity on it comes and goes depending on how much I
have been looking at such code recently. Right now I have not, but I
think we have most of the pieces of the puzzle. - We can offer codecs
based on the RAN side. We can choose the RAN side based on the codecs
offered in incoming SIP. We can HO from full to half-rate TCH inside the
same BTS, (i think) so I guess we can renegotiate the channel speech
mode. I don't think we can actually trigger SIP re-INVITE for codec
renegotiation. Maybe we can, I'm not sure what happens if we HO a FR TCH
to HR with that BTS defrag stuff.. Maybe we can't do that, I'd need to
check.
I'm still not sure what's in master in relation to late negotiation,
that is to say, we don't do channel assign and choose the RAN
speech_mode until we get SIP 200 back from the PBX with sdp confirming
the codec. That might be useful in the case where you don't establish
early media on the MO call and you wait until the B-leg of the PBX
answers and confirms codec. Let's say for example, you are going
upstream a a round-robin of VoIP providers, and some of them support
GSM-FR, so in the case you hit one, you want to use FR on the RAN and
not do any transcoding. If on the other hand upstream offers you only
PCMA/U, then maybe you choose to use AMR HR on the RAN site to conserve
capacity, or as you say, you call eventually goes to another GSM RAN and
so your B-leg in this case also maybe supports AMR.
Many possibilities, lot's of edge case stuff that I've long had full
intentions of implementing.. + The gods of time and energy often have
other plans
I'm actually not sure about the state of transcoding in the MGW, maybe
somebody has WIP in a branch. Don't think so though. I saw something the
other day about 3G to 2G, but I think it's "only" about the IuUP part. I
do need (partly for another task) to get a little more familiar with the
MGW code.
Point is: about these capabilities there is very
little
information available about what is supported and what is not within
the Osmocom domain.
Possibly/Probably because such things are in active
development.
k/