[calls] README: Add backend specific debugging information
- From: Evangelos Ribeiro Tzaras <devrtz src gnome org>
- To: commits-list gnome org
- Cc:
- Subject: [calls] README: Add backend specific debugging information
- Date: Mon, 22 Nov 2021 21:16:53 +0000 (UTC)
commit 748f9c937cee072b4f4c6509f1b32fd5cab704db
Author: Evangelos Ribeiro Tzaras <devrtz fortysixandtwo eu>
Date: Thu Nov 18 10:11:03 2021 +0100
README: Add backend specific debugging information
README.md | 28 ++++++++++++++++++++++++++++
1 file changed, 28 insertions(+)
---
diff --git a/README.md b/README.md
index 7310005e..6a955714 100644
--- a/README.md
+++ b/README.md
@@ -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)
useful.
+For backend specific debugging, please see the sections below.
+
## Call provider backends
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
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
@@ -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
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
This plugin is mostly useful for development purposes and work on the UI
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