README: Add backend specific debugging information

This commit is contained in:
Evangelos Ribeiro Tzaras
2021-11-18 10:11:03 +01:00
parent 5a2727c0f4
commit 748f9c937c

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@@ -50,6 +50,8 @@ If your system is using systemd you may find
[this guide](https://developer.puri.sm/Librem5/Development_Environment/Boards/Troubleshooting/Debugging.html) [this guide](https://developer.puri.sm/Librem5/Development_Environment/Boards/Troubleshooting/Debugging.html)
useful. useful.
For backend specific debugging, please see the sections below.
## Call provider backends ## Call provider backends
Calls uses libpeas to support runtime loadable plugins which we call "providers". Calls uses libpeas to support runtime loadable plugins which we call "providers".
@@ -83,6 +85,21 @@ This is the default backend for cellular calls. It uses `libmm-glib` to
talk to ModemManager over DBus. It currently only supports one modem and talk to ModemManager over DBus. It currently only supports one modem and
one active call at a time. one active call at a time.
#### Debugging
You can monitor the ModemManager messages on the DBus as follows:
gdbus monitor --system --dest org.freedesktop.ModemManager1
For complete debugging logs you can set ModemManager's log verbosity to DEBUG as follows:
mmcli -G DEBUG
and inspect the logs on a systemd based system with:
journalctl -u ModemManager.service
For more information see [here](https://modemmanager.org/docs/modemmanager/debugging/)
### SIP ### SIP
@@ -90,6 +107,17 @@ This plugin uses the libsofia-sip library for SIP signalling and
GStreamer for media handling. It supports multiple SIP accounts and GStreamer for media handling. It supports multiple SIP accounts and
currently one active call at a time (subject to change). currently one active call at a time (subject to change).
#### Debugging
You can print the sent and received SIP messages by setting the environment variable
`TPORT_LOG=1`. To test the audio quality you can use one of the various public
reachable echo test services, f.e. echo@conference.sip2sip.info. Please note that
the SIP plugin currently doesn't support DTMF, which is used for some test
services for navigating through a menu.
If one or both sides can't hear any audio at all it is likely that the audio
packets are not reaching the desired destination.
### Dummy ### Dummy
This plugin is mostly useful for development purposes and work on the UI This plugin is mostly useful for development purposes and work on the UI