Hi all!
Recently in the IRC I brought up a topic of testing voice calls in
OsmoCNI, which then turned into a discussion. The problem is I wanted
to highlight is that there is currently no *easy* way to test voice
calls, especially when running a virtual OsmoRAN setup (fake_trx.py +
trxcon or virtphy).
== A bit of history ==
Back in the openbsc/osmo-nitb days I used to run jolly's LCR (Linux Call
Router) [1], which was configured to play some hold melody when calling
995, and echo voice frames back when calling 999.
Nowadays openbsc/osmo-nitb is obsolete and completely suppressed by the
new and shiny OsmoRAN/OsmoCNI stack. The LCR is not actively maintained
anymore and does not support the MNCCv8, so it cannot talk to osmo-msc.
BTW, I forked it, fixed some compilation errors, created a package, and
tried to implement MNCCv8 support (no luck, calls still don't work):
https://gitea.osmocom.org/vyanitskiy/lcr/commits/branch/fixeria/fixeshttps://gitea.osmocom.org/vyanitskiy/lcr/commits/branch/fixeria/mncchttps://aur.archlinux.org/packages/lcr-git
Looks like it does not support late TCH assignment?
[1] http://www.linux-call-router.de/
== Current situation ==
Currently with the new post-NITB stack I see the following options:
a) run a virtual BTS, attach two mobiles, and setup a call between them;
b) run a real BTS, attach two phones, and setup a call between them;
c) run some PBX, talking to osmo-msc via osmo-sip-connector.
Personally I find neither of these options convenient because:
a) requires running two instances of the mobile app (from
osmocom-bb.git). I know one can run two and even more MS instances in
one mobile process, but this is still not handy.
b) requires running a real BTS and interacting with real phones. This
is what I usually do, but it takes more time than running everything
virtually.
c) requires setting up a PBX (e.g. Freeswitch, Asterisk), which in its
turn requires digging into the new world of configuration files. I do
have a repository with a know-to-work Freeswitch configuration [2], but
installing it (even from packages) is not trivial.
[2] https://people.osmocom.org/fixeria/freeswitch.cfg.zip
== What do I want ==
It would be great to have an easy-to-use echo service, be it attached to
osmo-msc via the MNCC socket, or be it built-in part of osmo-msc itself.
This would be usable for both real and virtual setups.
== Conclusion ==
I would like to know how do you guys test voice, and what do/would you
consider an easy approach. I actually found out that it's possible to
use osmo-msc's silent-call feature and play the Uplink RTP stream with
osmo-gapk. I'll share more details in a follow up mail.
In the IRC @whytek proposed to use sipp [3] and later came up with a
wiki page [4] describing how to achieve this with SEMS (SIP Express
Media Server). This is definitely easier than setting up Freeswitch or
Asterisk, but still feels like an overkill (sorry).
[3] https://sipp.sourceforge.net/doc/reference.html#RTP+echo
[4]
https://osmocom.org/projects/cellular-infrastructure/wiki/Simple_Echo_Server
Best regards,
Vadim.
--
- Vadim Yanitskiy <vyanitskiy at sysmocom.de> http://www.sysmocom.de/
=======================================================================
* sysmocom - systems for mobile communications GmbH
* Alt-Moabit 93
* 10559 Berlin, Germany
* Sitz / Registered office: Berlin, HRB 134158 B
* Geschaeftsfuehrer / Managing Director: Harald Welte
I'm also just trying to figure out how it works with global platform.
The problem you have with Global Platform is that you have not specified an
installation parameter. I tried it for example with "-default -params
80120100140600000000000000000000810600040200000400820600040200000400" and
the applet was installed and also made selectable.
You can find a discussion about that parameters here:
https://github.com/martinpaljak/GlobalPlatformPro/issues/293
I'm looking now why I can't open the menu e.g. from the iPhone. Probably
any access rules / directories access still need to be updated here?
