gajim issueshttps://dev.gajim.org/gajim/gajim/-/issues2018-08-23T19:38:20Zhttps://dev.gajim.org/gajim/gajim/-/issues/8308Jingle Audio/Video usage is unstable2018-08-23T19:38:20ZanonymousJingle Audio/Video usage is unstableSubject says it all.
"Toggle video session" button randomly becomes unavailable. For example, if some of peer's resources get offline (despite jingle-capable one stays online). Somebody (or both?) must disconnect & reconnect in order to...Subject says it all.
"Toggle video session" button randomly becomes unavailable. For example, if some of peer's resources get offline (despite jingle-capable one stays online). Somebody (or both?) must disconnect & reconnect in order to enable that button.
User who has update from 0.16 to -hg will have "ffmpegcolorspace" element in video_input_device thus his video calls will fail until he fixes it manually in config.
Then, if you start a call with peer, your peer sees your video in separate window. The windows layout at your peer is totally different from what you have. You have your chat window split in halves vertically, right side is black, left is text chat, in the middle there's minimized video of yourself (video test scheme in my case BTW, doesn't matter, just saying). Peer has no means to show you his video - his "toggle video" button is in "pressed" state.
And when the participant who has separate window with peer's video closes that window, his gajim crashes.
Mentions of "notify.py" make me think that it's related to lack of python3 port of python-notify lib, but having installed gajim without "libnotify" USE flag, i still get the same. Still may be issue of my -9999 ebuild, though.
$ gajim
/usr/lib64/python3.4/site-packages/gajim/common/gajim.py:204: PyGIWarning: GdkX11 was imported without specifying a version first. Use gi.require_version('GdkX11', '3.0') before import to ensure that the right version gets loaded.
from gi.repository import GdkX11
/usr/lib64/python3.4/site-packages/gajim/common/gajim.py:205: PyGIWarning: GstVideo was imported without specifying a version first. Use gi.require_version('GstVideo', '1.0') before import to ensure that the right version gets loaded.
from gi.repository import GstVideo
/usr/lib64/python3.4/site-packages/gajim/cell_renderer_image.py:33: PyGIDeprecationWarning: GObject.PARAM_READWRITE is deprecated; use GObject.ParamFlags.READWRITE (glib 2.42+) instead
'Image', GObject.PARAM_READWRITE),
/usr/lib64/python3.4/site-packages/gajim/statusicon.py:151: DeprecationWarning: Gtk.StatusIcon.set_visible is deprecated
self.status_icon.set_visible(True)
/usr/lib64/python3.4/site-packages/gajim/statusicon.py:140: DeprecationWarning: Gtk.StatusIcon.set_from_pixbuf is deprecated
self.status_icon.set_from_pixbuf(image.get_pixbuf())
/usr/lib64/python3.4/site-packages/gajim/roster_window.py:5088: DeprecationWarning: Gtk.StyleContext.get_background_color is deprecated
bgcolor = context.get_background_color(style)
19.03.2016 22:04:38 (E) gajim.c.ged Error while running an even handler: \<bound method RosterWindow._nec_roster_received of \<roster_window.RosterWindow object at 0x7fddd5a80978>>
Traceback (most recent call last):
File "/usr/lib64/python3.4/site-packages/gajim/common/ged.py", line 93, in raise_event
if handler(*args, **kwargs):
File "/usr/lib64/python3.4/site-packages/gajim/roster_window.py", line 2658, in _nec_roster_received
self.fire_up_unread_messages_events(obj.conn.name)
File "/usr/lib64/python3.4/site-packages/gajim/roster_window.py", line 1863, in fire_up_unread_messages_events
msg_log_id=result[0])
File "/usr/lib64/python3.4/site-packages/gajim/session.py", line 359, in roster_message
show_in_roster=obj.show_in_roster,
NameError: name 'obj' is not defined
/usr/lib64/python3.4/site-packages/gajim/conversation_textview.py:236: DeprecationWarning: Gtk.Widget.override_font is deprecated
self.tv.override_font(font)
/usr/lib64/python3.4/site-packages/gajim/chat_control.py:601: DeprecationWarning: Gtk.Widget.override_background_color is deprecated
banner_eventbox.override_background_color(Gtk.StateType.NORMAL, bg_color)
/usr/lib64/python3.4/site-packages/gajim/chat_control.py:604: DeprecationWarning: Gtk.Widget.override_color is deprecated
widget.override_color(Gtk.StateType.NORMAL, fg_color)
/usr/lib64/python3.4/site-packages/gajim/tooltips.py:365: DeprecationWarning: Gtk.Misc.set_alignment is deprecated
self.text_label.set_alignment(0, 0)
19.03.2016 22:04:40 (E) gajim.c.ged Error while running an even handler: \<bound method RosterWindow._nec_roster_received of \<roster_window.RosterWindow object at 0x7fddd5a80978>>
Traceback (most recent call last):
File "/usr/lib64/python3.4/site-packages/gajim/common/ged.py", line 93, in raise_event
if handler(*args, **kwargs):
File "/usr/lib64/python3.4/site-packages/gajim/roster_window.py", line 2658, in _nec_roster_received
self.fire_up_unread_messages_events(obj.conn.name)
File "/usr/lib64/python3.4/site-packages/gajim/roster_window.py", line 1863, in fire_up_unread_messages_events
msg_log_id=result[0])
File "/usr/lib64/python3.4/site-packages/gajim/session.py", line 359, in roster_message
show_in_roster=obj.show_in_roster,
NameError: name 'obj' is not defined
/usr/lib64/python3.4/site-packages/gajim/conversation_textview.py:709: DeprecationWarning: Gdk.Cursor.new is deprecated
w.set_cursor(Gdk.Cursor.new(Gdk.CursorType.HAND2))
19.03.2016 22:05:00 (E) gajim.c.ged Error while running an even handler: \<bound method Notification._nec_notification of \<notify.Notification object at 0x7fddd57e40b8>>
Traceback (most recent call last):
File "/usr/lib64/python3.4/site-packages/gajim/common/ged.py", line 93, in raise_event
if handler(*args, **kwargs):
File "/usr/lib64/python3.4/site-packages/gajim/notify.py", line 173, in _nec_notification
timeout=obj.popup_timeout)
File "/usr/lib64/python3.4/site-packages/gajim/notify.py", line 119, in popup
notification = Notify.Notification(_title, _text)
TypeError: GObject.__init__() takes exactly 0 arguments (2 given)
/usr/lib64/python3.4/site-packages/gajim/chat_control.py:1806: DeprecationWarning: Gtk.Image.set_from_stock is deprecated
Gtk.STOCK_NETWORK, 1)
19.03.2016 22:06:59 (E) gajim.c.ged Error while running an even handler: \<bound method Interface.handle_event_jingle_incoming of \<gui_interface.Interface object at 0x7fddd7fbfa58>>
Traceback (most recent call last):
File "/usr/lib64/python3.4/site-packages/gajim/common/ged.py", line 93, in raise_event
if handler(*args, **kwargs):
File "/usr/lib64/python3.4/site-packages/gajim/gui_interface.py", line 1286, in handle_event_jingle_incoming
path_to_image=path, title=event_type, text=txt)
File "/usr/lib64/python3.4/site-packages/gajim/notify.py", line 119, in popup
notification = Notify.Notification(_title, _text)
TypeError: GObject.__init__() takes exactly 0 arguments (2 given)
Segmentation fault
# Software versions
OS version:
GTK version:
PyGTK version:
# equery list \* | grep -i gtk
dev-python/pygtk-2.24.0-r4
dev-python/pygtksourceview-2.10.1-r1
dev-util/gtk-doc-1.24
dev-util/gtk-doc-am-1.24
dev-util/gtk-update-icon-cache-3.18.4
net-libs/gtk-vnc-0.5.4
net-libs/webkit-gtk-2.4.9
net-libs/webkit-gtk-2.10.7
net-misc/gtkvncviewer-0.4
net-misc/spice-gtk-0.30-r1
x11-libs/gtk+-2.24.29
x11-libs/gtk+-3.18.8
x11-libs/gtkglext-1.2.0-r3
x11-libs/gtksourceview-2.10.5-r3
x11-libs/wxGTK-3.0.2.0-r2https://dev.gajim.org/gajim/gajim/-/issues/8188Replace Audio/Video icons with standard icons2019-07-11T15:12:55ZDarlanReplace Audio/Video icons with standard icons# phenomenon
There are redundant icons that prevents from Gajim to integrate natively with system icon themes.
# background analysis
In `/usr/share/gajim/icons/hicolor/16x16/actions/` there are specific "inactive" icons such as gajim-cam...# phenomenon
There are redundant icons that prevents from Gajim to integrate natively with system icon themes.
# background analysis
In `/usr/share/gajim/icons/hicolor/16x16/actions/` there are specific "inactive" icons such as gajim-cam_inactive.png and gajim-mic_inactive.png.
# implementation recommendation
Use GTK+ toolkit to indicate on inactivity or unavailability of features. _Window Buttons_ of _Xfce4-panel_ turns icons to semi-transparent icons when a window is minimized.
Remove icons:
/actions/gajim-cam_active.png
/actions/gajim-cam_inactive.png
/actions/gajim-mic_active.png
/actions/gajim-mic_inactive.png
Set icons:
/devices/camera-web
/devices/audio-input-microphone1.2.0https://dev.gajim.org/gajim/gajim/-/issues/8108Video chat not available under Windows2018-05-25T09:13:22ZDarlanVideo chat not available under Windows# phenomenon
Outdated information.
# background analysis
Audio chat is available under Windows, video chat is not available under Windows.
# implementation recommendation
none
# phenomenon
Outdated information.
# background analysis
Audio chat is available under Windows, video chat is not available under Windows.
# implementation recommendation
none
https://dev.gajim.org/gajim/gajim/-/issues/8089E2E in audio-sessions -- SRTP in farstream2018-05-26T20:23:21ZanonymousE2E in audio-sessions -- SRTP in farstream= problem = End-to-End encryption in Gajim via ZRTP/DTLS
It seems like SRTP is now available in farstream -- is there still something missing to implement end-to-end encryption for audio-session (either by zrto or dtls)?
= problem = End-to-End encryption in Gajim via ZRTP/DTLS
It seems like SRTP is now available in farstream -- is there still something missing to implement end-to-end encryption for audio-session (either by zrto or dtls)?
https://dev.gajim.org/gajim/gajim/-/issues/7626Audio/Video devices not available in Preferences2019-09-14T11:43:13ZanonymousAudio/Video devices not available in Preferences# Bug description
I've used several other XMPP clients on this machine and each recognized sound card (IDT high definition audio Codec) and webcam (Dell Latitude E6410 integrated webcam), but not Gajim. The drop down boxes for sound and...# Bug description
I've used several other XMPP clients on this machine and each recognized sound card (IDT high definition audio Codec) and webcam (Dell Latitude E6410 integrated webcam), but not Gajim. The drop down boxes for sound and video are disabled.
Perhaps this is something that can be tweaked somewhere, then would you be able to provide guidance on what can be done for Gajim to discover that this laptop does have sound device and webcam?
Thank you!
# Steps to reproduce
Install Gajim on Latitude E6410 laptop
# Software versions
OS version: Windows 7 32 bit
GTK version: one that came with 0.15.4
PyGTK version: same as abovehttps://dev.gajim.org/gajim/gajim/-/issues/7580Add 'Start Audio/Video call' to contact context menu2021-09-03T23:19:48ZDarlanAdd 'Start Audio/Video call' to contact context menu= enhancement recommendation
Add VoIP context menu items.
|=Roster row=|=Menu item=|
|------------|-----------|
|Account|Start Chat...\\*Start Voice Chat...*\\Join Group Chat\\
|Contact|Start Chat\\*Start Voice Chat*\\Send Single Messag...= enhancement recommendation
Add VoIP context menu items.
|=Roster row=|=Menu item=|
|------------|-----------|
|Account|Start Chat...\\*Start Voice Chat...*\\Join Group Chat\\
|Contact|Start Chat\\*Start Voice Chat*\\Send Single Message...
Maybe *Start Voice Call_ or *Call to Contact_' would sound better.https://dev.gajim.org/gajim/gajim/-/issues/7564Crash when receive an incoming call2018-05-24T22:34:28ZanonymousCrash when receive an incoming call# Bug description
Gajim crash when I receive an incoming call
# Steps to reproduce
My friend calls with Gajim 0.16-alpha2-5ec89bc93747 and I have Gajim 0.16-5ec89bc93747.
He calls me by clicking on the Audio button in our conversation w...# Bug description
Gajim crash when I receive an incoming call
# Steps to reproduce
My friend calls with Gajim 0.16-alpha2-5ec89bc93747 and I have Gajim 0.16-5ec89bc93747.
He calls me by clicking on the Audio button in our conversation window.
I receive a windows to confirm if I want to speak with him.
When I click on « yes », gajim did not respond (←gajim ne répond pas)
# Software versions
OS version: me → Debian GNU/Linux testing (jessie) ; my friend → ArchLinux
GTK version: me and my friend → 2.24.22
PyGTK version: me and my friend → 2.24.0https://dev.gajim.org/gajim/gajim/-/issues/7464Make Audio/Video calls possible when using Zeroconf/Bonjour2022-01-21T08:55:13ZDarlanMake Audio/Video calls possible when using Zeroconf/Bonjour# Bug description
Gajim users can't communicate audio/video with Local account.
# Steps to reproduce
Open a chat with a local peer who is using Gajim and who have VoIP features enabled.
# Software versions
OS version: Salix OS 14.0.1 (...# Bug description
Gajim users can't communicate audio/video with Local account.
# Steps to reproduce
Open a chat with a local peer who is using Gajim and who have VoIP features enabled.
# Software versions
OS version: Salix OS 14.0.1 (Slackware 14.0 based)
GTK version: 2.24.10
PyGTK version: 2.24.0ThibgThibghttps://dev.gajim.org/gajim/gajim/-/issues/7303Error on try to accept an invitation to an audio call2018-08-23T19:52:16ZszpakError on try to accept an invitation to an audio callWhen I try to accept an invitation to an audio call I see following stack track on a console. In addition I don't have active audio/video icons in a chat dialog.
The other party uses Epiphany (Telepathy Gabble 0.16.1).
Traceback (most ...When I try to accept an invitation to an audio call I see following stack track on a console. In addition I don't have active audio/video icons in a chat dialog.
The other party uses Epiphany (Telepathy Gabble 0.16.1).
Traceback (most recent call last):
File "/usr/share/gajim/src/dialogs.py", line 5188, in on_voip_call_received_messagedialog_response
session.approve_content(content)
File "/usr/share/gajim/src/common/jingle_session.py", line 136, in approve_content
self.on_session_state_changed(content)
File "/usr/share/gajim/src/common/jingle_session.py", line 213, in on_session_state_changed
self.__session_accept()
File "/usr/share/gajim/src/common/jingle_session.py", line 614, in __session_accept
self.__broadcast(stanza, jingle, None, 'session-accept-sent')
File "/usr/share/gajim/src/common/jingle_session.py", line 474, in __broadcast
cn.on_stanza(stanza, content, error, action)
File "/usr/share/gajim/src/common/jingle_content.py", line 110, in on_stanza
callback(stanza, content, error, action)
File "/usr/share/gajim/src/common/jingle_content.py", line 91, in __on_content_accept
self.on_negotiated()
File "/usr/share/gajim/src/common/jingle_rtp.py", line 238, in on_negotiated
self.p2pstream.add_remote_candidates(self.transport.remote_candidates)
glib.GError: Invalid remote candidates passed, does not have the right username
ThibgThibghttps://dev.gajim.org/gajim/gajim/-/issues/7245Video does not work2018-08-23T20:11:33ZanonymousVideo does not work# Bug description
Audio works ok, videotest works ok, but it seems that Gajim is unable to get a video stream from my webcam. Other applications (Skype, Jitsi, Cheese, etc.) have no such problem.
After a contact accepts a videocall, noth...# Bug description
Audio works ok, videotest works ok, but it seems that Gajim is unable to get a video stream from my webcam. Other applications (Skype, Jitsi, Cheese, etc.) have no such problem.
After a contact accepts a videocall, nothing happens - no video window, no error message.
# Steps to reproduce
Make a video call.
In the Video Input Device combo box I have:
Autodetect,
V4L2: Default device,
V4L2: Philips SPC 1000NC Webcam,
Video test.
Nothing works, except Video test.
# Software versions
OS version:
On both sides Debian Squeeze.
Gajim 0.15 from Backports
https://dev.gajim.org/gajim/gajim/-/issues/7135Support for ZRTP (XEP-0262)2022-01-26T11:41:15ZzimioSupport for ZRTP (XEP-0262)XEP-0262: Use of ZRTP in Jingle RTP Sessions
http://xmpp.org/extensions/xep-0262.htmlXEP-0262: Use of ZRTP in Jingle RTP Sessions
http://xmpp.org/extensions/xep-0262.htmlThibgThibghttps://dev.gajim.org/gajim/gajim/-/issues/6117Audio/video does not work2018-08-23T20:01:29ZanonymousAudio/video does not work# Bug description
Initially it was reported in Fedora - https://bugzilla.redhat.com/show_bug.cgi?id=663335
I have tried http://mcepl.fedorapeople.org/rpms/gajim-0.15-0.1.20101220hg.1.fc15.noarch.rpm it seams closer to work. Buttons now ...# Bug description
Initially it was reported in Fedora - https://bugzilla.redhat.com/show_bug.cgi?id=663335
I have tried http://mcepl.fedorapeople.org/rpms/gajim-0.15-0.1.20101220hg.1.fc15.noarch.rpm it seams closer to work. Buttons now can be pressed, and request appeared on other side. In gajim banner appeared icon(s) on both side.
But when try accept call, in chat window only error to caller (translated):
Audio state : stop, reason: removed
# Steps to reproduce
Try start audio or video session.
# Software versions
OS version:
Linux Fedora 14
GTK version:
# rpm -q gtk2
PyGTK version:
# rpm -q pygtk2
pygtk2-2.17.0-7.fc14.i686ThibgThibghttps://dev.gajim.org/gajim/gajim/-/issues/6030No video on Windows2022-05-06T17:07:08ZanonymousNo video on Windows# Bug description
Although, I waited for the 14.1 update, but there seems to be no video avalaible.
# Steps to reproduce
Open a chat:The camera symbol is "grayed out".
# Software versions
OS version:
Windows Server 2008 R2 [x64], en, all...# Bug description
Although, I waited for the 14.1 update, but there seems to be no video avalaible.
# Steps to reproduce
Open a chat:The camera symbol is "grayed out".
# Software versions
OS version:
Windows Server 2008 R2 [x64], en, all patches
GTK version:
Sorry, don't know [I think, GTK comes with Gajim].
Note:This could also depend on this issue:#5928
br++mabrahttps://dev.gajim.org/gajim/gajim/-/issues/7555Diffie-Hellman parameters are required for forward secrecy2017-10-27T13:47:08Zfedor.brunnerDiffie-Hellman parameters are required for forward secrecy# Bug description
"To use perfect forward secrecy cipher suites, you must set up Diffie-Hellman parameters (on the server side), or the PFS cipher suites will be silently ignored."
The code for loading DH parameters is missing in jingl...# Bug description
"To use perfect forward secrecy cipher suites, you must set up Diffie-Hellman parameters (on the server side), or the PFS cipher suites will be silently ignored."
The code for loading DH parameters is missing in jingle_xtls.py
# Steps to reproduce
During file transfer using Jingle XTLS, only the cipher AES256-GCM-SHA384 is used, this cipher doesn't support PFS. (Note: to get information about cipher used in SSL connection the PyOpenSSL has to be patched, pyOpenSSL Bug 1249293)
# Fix
After this fix DHE-RSA-AES256-GCM-SHA384 will be used, this cipher supports PFS. The code tries to load user DH parameters from ~/.local/share/gajim/dh_params.pem, if this file doesn't exit (the user has not created his own DH parameters), the default application DH parameters will be loaded.
The default DH parameters can be downloaded from OpenSSL, please copy apps/dh4096.pem from OpenSSL to data/other/dh4096.pem so it's installed together with Gajim as default DH parameters.
## More security
It's recommended for security cautions user to create his own DH parameters and not use the default DH parameters, using command
openssl dhparam 4096 -out ~/.local/share/gajim/dh_params.pem
This command takes about 15minutes to complete. The user can also create DH parameters with more bits, but this takes much longer. There is no interface in pyOpenSSL to create DH parameters.
0.16ThibgThibghttps://dev.gajim.org/gajim/gajim/-/issues/5546jingle error cause account to disconnect2017-10-27T13:48:42Zmiscjingle error cause account to disconnect# Bug description
When I receive the following xml from empathy ( for example, when gajim crashed but the other side didn't and I chose to stop the video call, once gajim is back and connected ), gajim disconnect :
\<!-- In -->...# Bug description
When I receive the following xml from empathy ( for example, when gajim crashed but the other side didn't and I chose to stop the video call, once gajim is back and connected ), gajim disconnect :
\<!-- In -->
\<iq type="set" id="131851144379" to='misc@example.org/Portable/Akroma' from='phoebe@example.org/takara'>
\<jingle xmlns='urn:xmpp:jingle:1' action='session-info' sid='66' initiator='misc@example.org/Portable/Akroma'>
\<ringing xmlns='urn:xmpp:jingle:apps:rtp:info:1'/>
\</jingle>
\</iq>
\<!-- In -->
\<iq type="result" id="67" to='misc@example.org/Portable/Akroma' from='phoebe@example.org/takara'>
\</iq>
\<!-- Out -->
\<iq xmlns="urn:ietf:params:xml:ns:xmpp-stanzas" to="phoebe@example.org/takara" type="error" id="131851144379" from="misc@example.org/Portable/Akroma">
\<jingle action="session-info" initiator="misc@example.org/Portable/Akroma" xmlns="urn:xmpp:jingle:1" sid="66">
\<ringing xmlns="urn:xmpp:jingle:apps:rtp:info:1" />
\</jingle>
\<error code="500" type="cancel">
\<feature-not-implemented />
\</error>
\<unsupported-info xmlns="urn:xmpp:jingle:errors:1" />
\</iq>
\<!-- In -->
\<stream:error>
\<unsupported-stanza-type xmlns='urn:ietf:params:xml:ns:xmpp-streams'/>
\</stream:error>
\<!-- In -->
\</stream:stream>
misc@example is gajim hg, phoebe@example is running empathy 2.29. Both are on the same subnet.0.14ThibgThibghttps://dev.gajim.org/gajim/gajim/-/issues/7643Crashes on video start2018-12-25T12:32:46ZanonymousCrashes on video start# Bug description
When you start video chat Gajim crashes
# Steps to reproduce
Start Video call in gajim. It will crash with error ERROR:fs-rtp-discover-codecs.c:126:debug_codec_cap: assertion failed: (gst_caps_get_size (codec_cap->rtp_c...# Bug description
When you start video chat Gajim crashes
# Steps to reproduce
Start Video call in gajim. It will crash with error ERROR:fs-rtp-discover-codecs.c:126:debug_codec_cap: assertion failed: (gst_caps_get_size (codec_cap->rtp_caps) == 1)
latest farstream-0.1 0.1.2-3
Is the use of last farstream 0.2.3 possible?
# Software versions
OS version:
[[Linux 3.13.0-1 x86_64 GNU/Linux]]
GTK version:
PyGTK version:https://dev.gajim.org/gajim/gajim/-/issues/7611getaddrinfo throws exception2017-12-11T16:38:56Zfedor.brunnergetaddrinfo throws exception# Bug description
when there is no DNS record for the computer hostname, the socket.getaddrinfo fails with socket.gaierror exception
# Steps to reproduce
run Jingle file transfer with hostname which has not set up DNS record
...# Bug description
when there is no DNS record for the computer hostname, the socket.getaddrinfo fails with socket.gaierror exception
# Steps to reproduce
run Jingle file transfer with hostname which has not set up DNS record
Traceback (most recent call last):
File "/home/fedor/hg/gajim/src/dialogs.py", line 1490, in on_dialog_response
self.response_ok(dialog)
File "/home/fedor/hg/gajim/src/filetransfers_window.py", line 306, in on_ok
if self.send_file(account, contact, file_path, desc) \
File "/home/fedor/hg/gajim/src/filetransfers_window.py", line 361, in send_file
file_props)
File "/home/fedor/hg/gajim/src/common/jingle.py", line 168, in start_file_transfer
jingle.start_session()
File "/home/fedor/hg/gajim/src/common/jingle_session.py", line 291, in start_session
self.on_session_state_changed()
File "/home/fedor/hg/gajim/src/common/jingle_session.py", line 247, in on_session_state_changed
self.__session_initiate()
File "/home/fedor/hg/gajim/src/common/jingle_session.py", line 704, in __session_initiate
self.__broadcast(stanza, jingle, None, 'session-initiate-sent')
File "/home/fedor/hg/gajim/src/common/jingle_session.py", line 574, in __broadcast
cn.on_stanza(stanza, content, error, action)
File "/home/fedor/hg/gajim/src/common/jingle_content.py", line 114, in on_stanza
callback(stanza, content, error, action)
File "/home/fedor/hg/gajim/src/common/jingle_content.py", line 166, in __fill_jingle_stanza
content.addChild(node=self.transport.make_transport())
File "/home/fedor/hg/gajim/src/common/jingle_transport.py", line 129, in make_transport
self._add_local_ips_as_candidates()
File "/home/fedor/hg/gajim/src/common/jingle_transport.py", line 193, in _add_local_ips_as_candidates
for addr in socket.getaddrinfo(socket.gethostname(), None):
gaierror: [Errno -2] Name or service not known
0.16fedor.brunnerfedor.brunnerhttps://dev.gajim.org/gajim/gajim/-/issues/8811Audio/Video session throws errors.2018-08-27T15:57:00ZFlorian ApollonerAudio/Video session throws errors.## Versions
- OS: Debian unstable
- Gajim version: 0.98.2
- Python-nbxmpp version: 0.6.1
## Steps to reproduce the problem
1. Open a chat window, click on the hamburger and then `Audio session`/`Video session`
Error:
```
Tr...## Versions
- OS: Debian unstable
- Gajim version: 0.98.2
- Python-nbxmpp version: 0.6.1
## Steps to reproduce the problem
1. Open a chat window, click on the hamburger and then `Audio session`/`Video session`
Error:
```
Traceback (most recent call last):
File "/usr/lib/python3/dist-packages/gajim/chat_control.py", line 340, in _on_audio
self.on_jingle_button_toggled(state, 'audio')
File "/usr/lib/python3/dist-packages/gajim/chat_control.py", line 765, in on_jingle_button_toggled
'start_' + jingle_type)(self.contact.get_full_jid())
File "/usr/lib/python3/dist-packages/gajim/common/jingle.py", line 127, in start_audio
jingle.add_content('voice', JingleAudio(jingle))
File "/usr/lib/python3/dist-packages/gajim/common/jingle_rtp.py", line 336, in __init__
JingleRTPContent.__init__(self, session, 'audio', transport)
File "/usr/lib/python3/dist-packages/gajim/common/jingle_rtp.py", line 46, in __init__
JingleContent.__init__(self, session, transport)
TypeError: __init__() missing 1 required positional argument: 'senders'
```https://dev.gajim.org/gajim/gajim/-/issues/5785Video device not closed after Jingle calls2018-04-01T19:16:45ZFlorobVideo device not closed after Jingle calls# Bug description
I made a Jingle Video+Audio call to Gajim from my N900.
After closing the connection (again from the N900) the camera did not turn of. Upon trying to start another video call from Gajim it crashed (as in completely cras...# Bug description
I made a Jingle Video+Audio call to Gajim from my N900.
After closing the connection (again from the N900) the camera did not turn of. Upon trying to start another video call from Gajim it crashed (as in completely crashed no TB or anything).
# Software versions
OS version: Ubuntu 10.04
GTK+ version: 2.20.1
PyGTK version: 2.17.0ThibgThibghttps://dev.gajim.org/gajim/gajim/-/issues/6910Crash when send/receive voice/video calls2018-04-01T19:16:45ZanonymousCrash when send/receive voice/video calls# Bug description
Impossible to make voice/video calls.
Buttons are active, but after i press them - crash occures:
Traceback (most recent call last):
File "/usr/share/gajim/src/chat_control.py", line 1994, in on_video...# Bug description
Impossible to make voice/video calls.
Buttons are active, but after i press them - crash occures:
Traceback (most recent call last):
File "/usr/share/gajim/src/chat_control.py", line 1994, in on_video_button_toggled
self.on_jingle_button_toggled(widget, 'video')
File "/usr/share/gajim/src/chat_control.py", line 1982, in on_jingle_button_toggled
'start_' + jingle_type)(self.contact.get_full_jid())
File "/usr/share/gajim/src/common/jingle.py", line 123, in start_video
jingle.add_content('video', JingleVideo(jingle))
File "/usr/share/gajim/src/common/jingle_rtp.py", line 361, in __init__
self.setup_stream()
File "/usr/share/gajim/src/common/jingle_rtp.py", line 367, in setup_stream
JingleRTPContent.setup_stream(self)
File "/usr/share/gajim/src/common/jingle_rtp.py", line 78, in setup_stream
self.p2psession = self.conference.new_session(self.farsight_media)
GError: Could not create the rtp muxer element
Also tested incoming call from this bot - gabble.echo@test.collabora.co.uk
...also crash:
Traceback (most recent call last):
File "/usr/share/gajim/src/common/xmpp/idlequeue.py", line 530, in _process_events
return IdleQueue._process_events(self, fd, flags)
File "/usr/share/gajim/src/common/xmpp/idlequeue.py", line 400, in _process_events
obj.pollin()
File "/usr/share/gajim/src/common/xmpp/transports_nb.py", line 414, in pollin
self._do_receive()
File "/usr/share/gajim/src/common/xmpp/transports_nb.py", line 600, in _do_receive
self._on_receive(received)
File "/usr/share/gajim/src/common/xmpp/transports_nb.py", line 614, in _on_receive
self.on_receive(data)
File "/usr/share/gajim/src/common/xmpp/dispatcher_nb.py", line 452, in dispatch
handler['func'](session, stanza)
File "/usr/share/gajim/src/common/jingle.py", line 94, in _JingleCB
self.__sessions[sid].on_stanza(stanza)
File "/usr/share/gajim/src/common/jingle_session.py", line 299, in on_stanza
callable(stanza=stanza, jingle=jingle, error=error, action=action)
File "/usr/share/gajim/src/common/jingle_session.py", line 442, in __on_session_initiate
contents, contents_rejected, reason_txt = self.__parse_contents(jingle)
File "/usr/share/gajim/src/common/jingle_session.py", line 506, in __parse_contents
content = content_type(self, transport)
File "/usr/share/gajim/src/common/jingle_rtp.py", line 361, in __init__
self.setup_stream()
File "/usr/share/gajim/src/common/jingle_rtp.py", line 367, in setup_stream
JingleRTPContent.setup_stream(self)
File "/usr/share/gajim/src/common/jingle_rtp.py", line 78, in setup_stream
self.p2psession = self.conference.new_session(self.farsight_media)
GError: Could not create the rtp muxer element
These two crashes are for video calls. Voice calls reproduce almost the same crashes (difference in line numbers for several files)
# Steps to reproduce
just send/receive a call
# Software versions
OS version: Slackware64-current
GTK version: 2.24.4
PyGTK version: 2.22.00.15ThibgThibghttps://dev.gajim.org/gajim/gajim/-/issues/5545crash when accepting a video call2018-04-01T19:16:47Zmisccrash when accepting a video call# Bug description
Traceback (most recent call last):
File "/home/misc/checkout/hg/gajim/src/common/xmpp/idlequeue.py", line 528, in _process_events
return IdleQueue._process_events(self, fd, flags)
...# Bug description
Traceback (most recent call last):
File "/home/misc/checkout/hg/gajim/src/common/xmpp/idlequeue.py", line 528, in _process_events
return IdleQueue._process_events(self, fd, flags)
File "/home/misc/checkout/hg/gajim/src/common/xmpp/idlequeue.py", line 393, in _process_events
obj.pollin()
File "/home/misc/checkout/hg/gajim/src/common/xmpp/transports_nb.py", line 413, in pollin
self._do_receive()
File "/home/misc/checkout/hg/gajim/src/common/xmpp/transports_nb.py", line 599, in _do_receive
self._on_receive(received)
File "/home/misc/checkout/hg/gajim/src/common/xmpp/transports_nb.py", line 613, in _on_receive
self.on_receive(data)
File "/home/misc/checkout/hg/gajim/src/common/xmpp/dispatcher_nb.py", line 452, in dispatch
handler['func'](session, stanza)
File "/home/misc/checkout/hg/gajim/src/common/jingle.py", line 103, in _JingleCB
self.__sessions[(jid, sid)].on_stanza(stanza)
File "/home/misc/checkout/hg/gajim/src/common/jingle_session.py", line 294, in on_stanza
callable(stanza=stanza, jingle=jingle, error=error, action=action)
File "/home/misc/checkout/hg/gajim/src/common/jingle_session.py", line 429, in __on_session_initiate
contents, contents_rejected, reason = self.__parse_contents(jingle)
File "/home/misc/checkout/hg/gajim/src/common/jingle_session.py", line 488, in __parse_contents
content = content_type(self, transport)
File "/home/misc/checkout/hg/gajim/src/common/jingle_rtp.py", line 311, in __init__
self.setup_stream()
File "/home/misc/checkout/hg/gajim/src/common/jingle_rtp.py", line 317, in setup_stream
JingleRTPContent.setup_stream(self)
File "/home/misc/checkout/hg/gajim/src/common/jingle_rtp.py", line 77, in setup_stream
if not stun_server and self._stun_servers:
AttributeError: 'JingleVideo' object has no attribute '_stun_servers'
I tried to establish a video call between empathy and gajim latest hg checkout.
indeed, from looking at the code, the stun_server is stored elsehwere. Here is a patch that solve the problem for me.
# Steps to reproduce
- accept video call
0.14ThibgThibghttps://dev.gajim.org/gajim/gajim/-/issues/8210Programming error on initiating a Jingle session2018-04-29T20:37:30ZDarlanProgramming error on initiating a Jingle session# Bug description
Traceback (most recent call last):
File "/usr/lib/python2.7/site-packages/nbxmpp/dispatcher_nb.py", line 495, in dispatch
handler['func'](session, stanza)
File "/usr/share/gajim/...# Bug description
Traceback (most recent call last):
File "/usr/lib/python2.7/site-packages/nbxmpp/dispatcher_nb.py", line 495, in dispatch
handler['func'](session, stanza)
File "/usr/share/gajim/src/common/jingle.py", line 109, in _JingleCB
self._sessions[sid].on_stanza(stanza)
File "/usr/share/gajim/src/common/jingle_session.py", line 343, in on_stanza
call(stanza=stanza, jingle=jingle, error=error, action=action)
File "/usr/share/gajim/src/common/jingle_session.py", line 574, in __broadcast
cn.on_stanza(stanza, content, error, action)
File "/usr/share/gajim/src/common/jingle_content.py", line 117, in on_stanza
callback(stanza, content, error, action)
File "/usr/share/gajim/src/common/jingle_rtp.py", line 285, in __on_remote_codecs
self.p2pstream.set_remote_codecs(codecs)
GError: There was no intersection between the remote codecs and the local ones
# Steps to reproduce
1. Enter to an MUC whee JIDs are _not_ public to anyone.
2. Remote-end (room admin or owner), which is not in roster, attempts to initiate a Jingle session using Jitsi 2.8.build.2.8.5426 for Linux.
# Software versions
Gajim version: 0.16.5-37b54cc9488f
OS version: Salix OS 14.1 (Slackware 14.1 based)
GTK version: 2.24.20
PyGTK version: 2.24.00.16.6Yann LeboulangerYann Leboulangerhttps://dev.gajim.org/gajim/gajim/-/issues/8191Gajim 2 Gajim File Transfer doesn't work with IPv6 enabled but not used (Patc...2018-04-29T20:37:30ZanonymousGajim 2 Gajim File Transfer doesn't work with IPv6 enabled but not used (Patch attached)# Bug description
If IPv6 is enabled but not activly used, Gajim sends its peer a ::1 address for file transfer.
# Steps to reproduce
Enable ipv6 in Linux kernel, try to send a file to another linux user with p2p.
# Software versions
O...# Bug description
If IPv6 is enabled but not activly used, Gajim sends its peer a ::1 address for file transfer.
# Steps to reproduce
Enable ipv6 in Linux kernel, try to send a file to another linux user with p2p.
# Software versions
OS version:
GTK version:
PyGTK version:0.16.5https://dev.gajim.org/gajim/gajim/-/issues/7713Jingle File Transfer doesn't work without Farstream library2018-04-29T20:37:45Zfedor.brunnerJingle File Transfer doesn't work without Farstream library# Bug description
Jingle File Transfers don't work if Farstream library is not installed. But the implementation of Jingle File Transfers doesn't depend on Farstream.
# Steps to reproduce
Install Gajim without Farstream and try to do ...# Bug description
Jingle File Transfers don't work if Farstream library is not installed. But the implementation of Jingle File Transfers doesn't depend on Farstream.
# Steps to reproduce
Install Gajim without Farstream and try to do a file transfer, only SI file transfer is possible.
[https://trac.gajim.org/wiki/JingleFileTransfer]
# Software versions
Gajim 0.16 hg0.16fedor.brunnerfedor.brunnerhttps://dev.gajim.org/gajim/gajim/-/issues/7541Prefer stronger hash algorithms2018-04-29T20:37:50Zfedor.brunnerPrefer stronger hash algorithms# Bug description
During Jingle transfer the weak hash function MD5 is used even when the contact supports strong hash function SHA2-512. This is because of a bug in `__hash_support` in jingle.py
The current code is:
def ...# Bug description
During Jingle transfer the weak hash function MD5 is used even when the contact supports strong hash function SHA2-512. This is because of a bug in `__hash_support` in jingle.py
The current code is:
def __hash_support(self, contact):
if contact.supports(nbxmpp.NS_HASHES):
if contact.supports(nbxmpp.NS_HASHES_MD5):
return 'md5'
elif contact.supports(nbxmpp.NS_HASHES_SHA1):
return 'sha-1'
elif contact.supports(nbxmpp.NS_HASHES_SHA256):
return 'sha-256'
elif contact.supports(nbxmpp.NS_HASHES_SHA512):
return 'sha-512'
return None
The correct code should be:
def __hash_support(self, contact):
if contact.supports(nbxmpp.NS_HASHES):
if contact.supports(nbxmpp.NS_HASHES_SHA512):
return 'sha-512'
elif contact.supports(nbxmpp.NS_HASHES_SHA256):
return 'sha-256'
elif contact.supports(nbxmpp.NS_HASHES_SHA1):
return 'sha-1'
elif contact.supports(nbxmpp.NS_HASHES_MD5):
return 'md5'
return None
# Steps to reproduce
Can be reproduced with Jingle file transfer0.16ThibgThibghttps://dev.gajim.org/gajim/gajim/-/issues/6999AttributeError: 'JingleTransportIBB' object has no attribute 'set_connection'2018-04-29T20:38:15ZZashAttributeError: 'JingleTransportIBB' object has no attribute 'set_connection'# Bug description
Traceback (most recent call last):
File "/home/zash/src/gajim/src/common/xmpp/idlequeue.py", line 533, in _process_events
return IdleQueue._process_events(self, fd, flags)
File "...# Bug description
Traceback (most recent call last):
File "/home/zash/src/gajim/src/common/xmpp/idlequeue.py", line 533, in _process_events
return IdleQueue._process_events(self, fd, flags)
File "/home/zash/src/gajim/src/common/xmpp/idlequeue.py", line 394, in _process_events
obj.pollin()
File "/home/zash/src/gajim/src/common/xmpp/transports_nb.py", line 414, in pollin
self._do_receive()
File "/home/zash/src/gajim/src/common/xmpp/transports_nb.py", line 600, in _do_receive
self._on_receive(received)
File "/home/zash/src/gajim/src/common/xmpp/transports_nb.py", line 614, in _on_receive
self.on_receive(data)
File "/home/zash/src/gajim/src/common/xmpp/dispatcher_nb.py", line 453, in dispatch
handler['func'](session, stanza)
File "/home/zash/src/gajim/src/common/jingle.py", line 108, in _JingleCB
self._sessions[sid].on_stanza(stanza)
File "/home/zash/src/gajim/src/common/jingle_session.py", line 328, in on_stanza
callable(stanza=stanza, jingle=jingle, error=error, action=action)
File "/home/zash/src/gajim/src/common/jingle_session.py", line 498, in __on_session_initiate
contents, contents_rejected, reason_txt = self.__parse_contents(jingle)
File "/home/zash/src/gajim/src/common/jingle_session.py", line 576, in __parse_contents
content = content_type(self, transport)
File "/home/zash/src/gajim/src/common/jingle_ft.py", line 91, in __init__
self.transport.set_connection(session.connection)
AttributeError: 'JingleTransportIBB' object has no attribute 'set_connection'
# Steps to reproduce
Receive this:
\<iq id="8" type="set">
\<jingle xmlns="urn:xmpp:jingle:1" sid="171771a3-07a3-4448-8bc5-6703287bcd86" initiator="" action="session-initiate">
\<content creator="initiator" name="file">
\<description xmlns="urn:xmpp:jingle:apps:file-transfer:3">
\<offer>
\<file xmlns="http://jabber.org/protocol/si/profile/file-transfer" name="jingle.txt" size="6144">
\<desc/>
\</file>
\</offer>
\</description>
\<transport xmlns="urn:xmpp:jingle:transports:ibb:1" block-size="2048" stanza="iq" sid="c37a5783-2e05-42c3-bfe3-50861a1df61b"/>
\</content>
\</jingle>
\</iq>
# Software versions
rfa253547651b on jingleFT branchzimiozimiohttps://dev.gajim.org/gajim/gajim/-/issues/6998Jingle FT with clients only supporting IBB2018-04-29T20:38:15ZZashJingle FT with clients only supporting IBB# Bug description
Sending a file to a client with only IBB, results in Gajim offering socks5 transport.
= Steps to reproduce =2
Send file to a client only advertising support for the IBB transport:
\<feature var='urn:xmpp:jin...# Bug description
Sending a file to a client with only IBB, results in Gajim offering socks5 transport.
= Steps to reproduce =2
Send file to a client only advertising support for the IBB transport:
\<feature var='urn:xmpp:jingle:transports:ibb:1'/>
# Software versions
rfa253547651b (jingleFT branch)0.15zimiozimiohttps://dev.gajim.org/gajim/gajim/-/issues/6982Initiating audio causes error message2018-04-29T20:38:15ZanonymousInitiating audio causes error message# Bug description
When clicking the microphone button to initiate an audio conversation, I receive the following dialog box:
Error: GStreamer 上で一般的なリソースエラーが起きました (translation: there was a resource error in GStreamer)
Debug: pulsesink....# Bug description
When clicking the microphone button to initiate an audio conversation, I receive the following dialog box:
Error: GStreamer 上で一般的なリソースエラーが起きました (translation: there was a resource error in GStreamer)
Debug: pulsesink.c(549): gst_pulseringbuffer_open_device (): /GstPulseSink:autoaudiosink0-actual-sink-pulse
# Steps to reproduce
Click the microphone button in a chat window.
# Software versions
OS version:
[[Debian Wheezy (Testing)]]
GTK version:
[[2.24.4]]
PyGTK version: 2.24.0
Note: I am using ALSA, not PulseAudio.ThibgThibghttps://dev.gajim.org/gajim/gajim/-/issues/6894cannot make a call from Gajim to Maemo native client2018-04-29T20:38:22Zanonymouscannot make a call from Gajim to Maemo native client# Bug description
cannot make a call from Gajim to Maemo native client
# Steps to reproduce
make a call to someone using Maemo operative system with XMPP native client
# Software versions
OS version: Debian Squeeze
GTK version: 2.20....# Bug description
cannot make a call from Gajim to Maemo native client
# Steps to reproduce
make a call to someone using Maemo operative system with XMPP native client
# Software versions
OS version: Debian Squeeze
GTK version: 2.20.1
PyGTK version: 2.17.0
Im on Gajim 0.14.1ThibgThibghttps://dev.gajim.org/gajim/gajim/-/issues/6859Voice calls - pulseaudio - chopping fix2018-04-29T20:38:23ZanonymousVoice calls - pulseaudio - chopping fixHello,
I have sent you recently patch[1] which added support for pulseaudio in voice calls.
However I found out that sound is chopping after couple of minutes of call and usually sound is lost completely after couple of other minutes....Hello,
I have sent you recently patch[1] which added support for pulseaudio in voice calls.
However I found out that sound is chopping after couple of minutes of call and usually sound is lost completely after couple of other minutes.
This doesn't happen once sync is enabled for pulsesink. Please find the patch attached.
Thanks for adding to your sources!
Tomas
[1]
http://www.lagaule.org/pipermail/gajim-devel/2011-March/000767.html
0.15ThibgThibghttps://dev.gajim.org/gajim/gajim/-/issues/6780cannot make call to a GTalk user2018-04-29T20:38:25Zanonymouscannot make call to a GTalk user# Bug description
Cannot call a user who is using GTalk, although he can make calls to users but not me. Also I can call a user using PSI 0.14
# Steps to reproduce
Try to call a user w/ a standart GTalk client.
# Software versions
OS v...# Bug description
Cannot call a user who is using GTalk, although he can make calls to users but not me. Also I can call a user using PSI 0.14
# Steps to reproduce
Try to call a user w/ a standart GTalk client.
# Software versions
OS version:
Debian sid
Gajim version:
0.14.1ThibgThibghttps://dev.gajim.org/gajim/gajim/-/issues/6114Can't hang up voice with user2018-04-29T20:38:27ZanonymousCan't hang up voice with user# Bug description
There is no possibility to hang up the call if user has no voice capability. I mean if gajim thinks that user has no voice support but this user calls to it and call negotiated I will not be able to hang up the call bec...# Bug description
There is no possibility to hang up the call if user has no voice capability. I mean if gajim thinks that user has no voice support but this user calls to it and call negotiated I will not be able to hang up the call because the button is inactive.
# Steps to reproduce
1. Call from jid with, for example, subscription "from" then we can't know user's capabilities
2. Accept the call
3. Can't hang up from gajim
# Software versions
OS version:
Gentoo
GTK version:
PyGTK version:0.15ThibgThibghttps://dev.gajim.org/gajim/gajim/-/issues/6108ICE transport gives wrong port number2018-04-29T20:38:27ZanonymousICE transport gives wrong port number# Bug description
If client is behind NAT, Gajim uses STUN server to determine public IP address and port which can be used to connect. But Gajim sends transport in such manner:
\<candidate foundation="1" protocol="udp" network...# Bug description
If client is behind NAT, Gajim uses STUN server to determine public IP address and port which can be used to connect. But Gajim sends transport in such manner:
\<candidate foundation="1" protocol="udp" network="0" generation="0" ip="217.25.221.127" component="1" priority="1677721855" type="srflx" port="40129" />
And it transmit local port not remote. It must use remote port and put local port into rel-port attribute as described in XEP-176.
This bug makes gajim voice support unusable behind restrict NAT.
# Steps to reproduce
# Software versions
OS version:
GTK version:
PyGTK version:ThibgThibghttps://dev.gajim.org/gajim/gajim/-/issues/6098Failed to make a voice call twice2018-04-29T20:38:29ZanonymousFailed to make a voice call twice# Bug description
If I try to call (using psi+ or pidgin), hang up and then call again then I see this exception:
Error: Could not decode stream.
Debug: gstsirendec.c(317): gst_siren_dec_chain (): /GstPipeline:pipeline3/FsRtpConference:f...# Bug description
If I try to call (using psi+ or pidgin), hang up and then call again then I see this exception:
Error: Could not decode stream.
Debug: gstsirendec.c(317): gst_siren_dec_chain (): /GstPipeline:pipeline3/FsRtpConference:fsrtpconference3/GstBin:recv_1_4013730427_96/GstSirenDec:sirendec1:
Error decoding frame: 7
# Steps to reproduce
1. Call to Psi+ or Pidgin
2. Talk
3. Hang up
4. Call again
5. Exception
# Software versions
OS version:
Gentoo
GTK version:
2.20.1-r1
PyGTK version:
2.17.0ThibgThibghttps://dev.gajim.org/gajim/gajim/-/issues/6046Can not start the audio / video session2018-04-29T20:38:30ZDicsonCan not start the audio / video session# Bug description
I can initiate an audio / video session just after my companion reconnects
# Steps to reproduce
1 Start Gajim
2 start chat (audio / video buttons is inactive)
3 my companion reconnect - audio / video buttons is active
# Bug description
I can initiate an audio / video session just after my companion reconnects
# Steps to reproduce
1 Start Gajim
2 start chat (audio / video buttons is inactive)
3 my companion reconnect - audio / video buttons is active
0.15ThibgThibghttps://dev.gajim.org/gajim/gajim/-/issues/5919TB while attempting to make voicecall.2018-04-29T20:38:37ZanonymousTB while attempting to make voicecall.Was testing jingle with bots
echo@ haar.student.utwente.nl
echo@ test.collabora.co.uk
The second was ok, but I couldn't establish audio session with
echo@ haar.student.utwente.nl
Traceback (most recent call last):
...Was testing jingle with bots
echo@ haar.student.utwente.nl
echo@ test.collabora.co.uk
The second was ok, but I couldn't establish audio session with
echo@ haar.student.utwente.nl
Traceback (most recent call last):
File "/usr/share/gajim/src/common/xmpp/idlequeue.py", line 528, in _process_events
return IdleQueue._process_events(self, fd, flags)
File "/usr/share/gajim/src/common/xmpp/idlequeue.py", line 393, in _process_events
obj.pollin()
File "/usr/share/gajim/src/common/xmpp/transports_nb.py", line 414, in pollin
self._do_receive()
File "/usr/share/gajim/src/common/xmpp/transports_nb.py", line 600, in _do_receive
self._on_receive(received)
File "/usr/share/gajim/src/common/xmpp/transports_nb.py", line 614, in _on_receive
self.on_receive(data)
File "/usr/share/gajim/src/common/xmpp/dispatcher_nb.py", line 452, in dispatch
handler['func'](session, stanza)
File "/usr/share/gajim/src/common/jingle.py", line 94, in _JingleCB
self.__sessions[sid].on_stanza(stanza)
File "/usr/share/gajim/src/common/jingle_session.py", line 299, in on_stanza
callable(stanza=stanza, jingle=jingle, error=error, action=action)
File "/usr/share/gajim/src/common/jingle_session.py", line 472, in __broadcast
cn.on_stanza(stanza, content, error, action)
File "/usr/share/gajim/src/common/jingle_content.py", line 103, in on_stanza
callback(stanza, content, error, action)
File "/usr/share/gajim/src/common/jingle_rtp.py", line 254, in __on_remote_codecs
self.farsight_media, int(codec['clockrate']))
TypeError: int() argument must be a string or a number, not 'NoneType'
Ubuntu 10.04. Gajim 0.14 from the PPA.0.14.1ThibgThibghttps://dev.gajim.org/gajim/gajim/-/issues/5881Jingle doesn't work2018-04-29T20:38:39ZanonymousJingle doesn't work# Bug description
When I click microphone button gajim prints in terminal:
Traceback (most recent call last):
File "/usr/local/share/gajim/src/chat_control.py", line 1962, in on_audio_button_toggled
self....# Bug description
When I click microphone button gajim prints in terminal:
Traceback (most recent call last):
File "/usr/local/share/gajim/src/chat_control.py", line 1962, in on_audio_button_toggled
self.on_jingle_button_toggled(widget, 'audio')
File "/usr/local/share/gajim/src/chat_control.py", line 1953, in on_jingle_button_toggled
'start_' + jingle_type)(self.contact.get_full_jid())
File "/usr/local/share/gajim/src/common/jingle.py", line 110, in start_audio
jingle.add_content('voice', JingleAudio(jingle))
File "/usr/local/share/gajim/src/common/jingle_rtp.py", line 304, in __init__
self.setup_stream()
File "/usr/local/share/gajim/src/common/jingle_rtp.py", line 319, in setup_stream
JingleRTPContent.setup_stream(self)
File "/usr/local/share/gajim/src/common/jingle_rtp.py", line 78, in setup_stream
self.p2psession = self.conference.new_session(self.farsight_media)
glib.GError: Unknown error while trying to discover codecs
And when I click again it prints:
Traceback (most recent call last):
File "/usr/local/share/gajim/src/chat_control.py", line 1962, in on_audio_button_toggled
self.on_jingle_button_toggled(widget, 'audio')
File "/usr/local/share/gajim/src/chat_control.py", line 1953, in on_jingle_button_toggled
'start_' + jingle_type)(self.contact.get_full_jid())
File "/usr/local/share/gajim/src/common/jingle.py", line 110, in start_audio
jingle.add_content('voice', JingleAudio(jingle))
File "/usr/local/share/gajim/src/common/jingle_rtp.py", line 304, in __init__
self.setup_stream()
File "/usr/local/share/gajim/src/common/jingle_rtp.py", line 319, in setup_stream
JingleRTPContent.setup_stream(self)
File "/usr/local/share/gajim/src/common/jingle_rtp.py", line 78, in setup_stream
self.p2psession = self.conference.new_session(self.farsight_media)
glib.GError: No codecs for media type audio detected
Further clicks cause repeat of this message.
# Steps to reproduce
Find someone with jingle support and try to make voice call.
# Software versions
OS version:
OpenSuSE 11.3
GTK version:
2.20.1
PyGTK version:
2.17.0
GStreamer, its plugins (base, good, bad, ugly), Farsight, python farsight, gstreamer farsight, libnice. Everything is fresh enough. How can I get those needed codecs (at least which exactly are needed)?ThibgThibghttps://dev.gajim.org/gajim/gajim/-/issues/5878Video doesn't show up2018-04-29T20:38:39ZanonymousVideo doesn't show up# Bug description
I compiled the new version 0.14 on Ubuntu Lucid Lynx. Everything looks ok, help says i really got version 0.14, i can chat and i can make "audio-chats" with the test-account echoATtest.collabora.co.uk This echo contact...# Bug description
I compiled the new version 0.14 on Ubuntu Lucid Lynx. Everything looks ok, help says i really got version 0.14, i can chat and i can make "audio-chats" with the test-account echoATtest.collabora.co.uk This echo contacts sends everything back to you what you send. So it answers chats, you hear what you say via xmpp/jingle and it sends also your video back.
The video is the problem. I can start the video-feed and i can see that the status led of my webcam turns blue, but after that nothing happens. I don't see my video nor i can see the video which should be sent back to me. I checked echoATtest.collabora.co.uk with Empathy and the same account, i can do audio-/video-chats with empathy...
# Steps to reproduce
Get gajim 0.14. Add echo@test.collabora.co.uk as a contact, start a video-chat and see what happens. I can see that the led of my webcam turns on, but i don't see a video of myself nor the video which is sent back to me. You can see a log at
http://paste.ubuntuusers.de/398875/
# Software versions
OS version:
Ubuntu Linux 10.04
GTK version:
GTK 2.20
ThibgThibghttps://dev.gajim.org/gajim/gajim/-/issues/5794Jingle errors should be handled/displayed according to severity2018-04-29T20:38:44ZFlorobJingle errors should be handled/displayed according to severityCurrently all jingle error conditions are displayed the same way (Green text in the chat dialogue). Behaviour should be different for different types. E.g. an unsupported info should probably not cause output. Telling the user that you d...Currently all jingle error conditions are displayed the same way (Green text in the chat dialogue). Behaviour should be different for different types. E.g. an unsupported info should probably not cause output. Telling the user that you don't know what something purely informational means is at best confusing.0.14ThibgThibghttps://dev.gajim.org/gajim/gajim/-/issues/5793Error when closing video window2018-04-29T20:38:44ZZashError when closing video windowIf I start a jingle video session and then close the video output window, I get this popup:
GStreamer error
Error: Resource not found.
Debug: xvimagesink.c(1297): gst_xvimagesink_handle_xevents (): /Gst...If I start a jingle video session and then close the video output window, I get this popup:
GStreamer error
Error: Resource not found.
Debug: xvimagesink.c(1297): gst_xvimagesink_handle_xevents (): /GstPipeline:pipeline8/GstBin:bin63/GstAutoVideoSink:autovideosink6/GstXvImageSink:autovideosink6-actual-sink-xvimage
The other end get gstreamers test image.0.16ThibgThibghttps://dev.gajim.org/gajim/gajim/-/issues/5777Voice calls aren't availible but possible2018-04-29T20:38:44ZanonymousVoice calls aren't availible but possibleI have Gajim from a mercurial (fresh version) under Ubuntu 10.04 32-bit on the one side and Psi+ 0.15.2460 under Windows XP on the other side. It isn't possible to call from Gajim to Psi simply because "Audio call" button in the chat win...I have Gajim from a mercurial (fresh version) under Ubuntu 10.04 32-bit on the one side and Psi+ 0.15.2460 under Windows XP on the other side. It isn't possible to call from Gajim to Psi simply because "Audio call" button in the chat window is not clickable. But it is possible to call from Psi to Gajim and then I can talk without any problems, except I can't end a call from the Gajim side because "Audio call" button isn't clickable too.ThibgThibghttps://dev.gajim.org/gajim/gajim/-/issues/5662gajim doesn't starts single sided video stream if the other side doesn't have...2018-04-29T20:38:51Zanonymousgajim doesn't starts single sided video stream if the other side doesn't have a cam...# Bug description
gajim doesn't starts single sided video stream if the other side doesn't have a cam.
after starting video conference it just happens nothing. not even a error message.
but after setting the side without a cam to video t...# Bug description
gajim doesn't starts single sided video stream if the other side doesn't have a cam.
after starting video conference it just happens nothing. not even a error message.
but after setting the side without a cam to video test source in the preferences, the single sided video works
# Software versions
OS version:
GTK version:
PyGTK version:ThibgThibghttps://dev.gajim.org/gajim/gajim/-/issues/5653Error when someone is using Lampiro2018-04-29T20:38:51ZanonymousError when someone is using Lampiro# Bug description
My friend uses Lampiro (http://lampiro.bluendo.com) on his cellphone. Unfortunately something with this client triggers Traceback in Gajim:
Traceback (most recent call last):
File "/usr/share/gajim/...# Bug description
My friend uses Lampiro (http://lampiro.bluendo.com) on his cellphone. Unfortunately something with this client triggers Traceback in Gajim:
Traceback (most recent call last):
File "/usr/share/gajim/src/common/xmpp/idlequeue.py", line 528, in _process_events
return IdleQueue._process_events(self, fd, flags)
File "/usr/share/gajim/src/common/xmpp/idlequeue.py", line 393, in _process_events
obj.pollin()
File "/usr/share/gajim/src/common/xmpp/transports_nb.py", line 413, in pollin
self._do_receive()
File "/usr/share/gajim/src/common/xmpp/transports_nb.py", line 599, in _do_receive
self._on_receive(received)
File "/usr/share/gajim/src/common/xmpp/transports_nb.py", line 613, in _on_receive
self.on_receive(data)
File "/usr/share/gajim/src/common/xmpp/dispatcher_nb.py", line 452, in dispatch
handler['func'](session, stanza)
File "/usr/share/gajim/src/common/jingle.py", line 103, in _JingleCB
self.__sessions[(jid, sid)].on_stanza(stanza)
File "/usr/share/gajim/src/common/jingle_session.py", line 294, in on_stanza
callable(stanza=stanza, jingle=jingle, error=error, action=action)
File "/usr/share/gajim/src/common/jingle_session.py", line 429, in __on_session_initiate
contents, contents_rejected, reason = self.__parse_contents(jingle)
File "/usr/share/gajim/src/common/jingle_session.py", line 499, in __parse_contents
failed.add('unsupported-applications')
NameError: global name 'failed' is not defined
It appears that this problem exists in jingle component.
This traceback showed up when my friend logged in with Lampiro for the first time and then again when he tried to send me a file.
# Steps to reproduce
Have a friend using Lampiro ;)
# Software versions
OS version: Gentoo Linux
GTK version: 2.18.7
PyGTK version: 2.16.00.14ThibgThibghttps://dev.gajim.org/gajim/gajim/-/issues/5651TB while trying to accept a Jingle voice call2018-04-29T20:38:51ZanonymousTB while trying to accept a Jingle voice call# Bug description
\<pre>
Traceback (most recent call last):
File "/USR3/src/gajim/gajim/src/common/xmpp/idlequeue.py", line 528, in _process_events
return IdleQueue._process_events(self, fd, flags)
File "/USR3/src/gajim/gajim/sr...# Bug description
\<pre>
Traceback (most recent call last):
File "/USR3/src/gajim/gajim/src/common/xmpp/idlequeue.py", line 528, in _process_events
return IdleQueue._process_events(self, fd, flags)
File "/USR3/src/gajim/gajim/src/common/xmpp/idlequeue.py", line 393, in _process_events
obj.pollin()
File "/USR3/src/gajim/gajim/src/common/xmpp/transports_nb.py", line 413, in pollin
self._do_receive()
File "/USR3/src/gajim/gajim/src/common/xmpp/transports_nb.py", line 599, in _do_receive
self._on_receive(received)
File "/USR3/src/gajim/gajim/src/common/xmpp/transports_nb.py", line 613, in _on_receive
self.on_receive(data)
File "/USR3/src/gajim/gajim/src/common/xmpp/dispatcher_nb.py", line 452, in dispatch
handler['func'](session, stanza)
File "/USR3/src/gajim/gajim/src/common/jingle.py", line 103, in _JingleCB
self.__sessions[(jid, sid)].on_stanza(stanza)
File "/USR3/src/gajim/gajim/src/common/jingle_session.py", line 294, in on_stanza
callable(stanza=stanza, jingle=jingle, error=error, action=action)
File "/USR3/src/gajim/gajim/src/common/jingle_session.py", line 453, in __broadcast
cn = self.contents[(creator, name)]
KeyError: (u'initiator', u'stream1')
\</pre>
# Steps to reproduce
Add echo@test.collabora.co.uk to the roster, send "!playsong" to it.
# Software versions
OS version: Debian unstable
GTK version: 2.18.7-1
PyGTK version: 2.16.0-20.14https://dev.gajim.org/gajim/gajim/-/issues/5525Bug when starting a video or audio call2018-04-29T20:38:58ZanonymousBug when starting a video or audio call# Bug description
Traceback (most recent call last):
File "/usr/share/gajim/src/common/jingle_rtp.py", line 178, in _on_gst_message
self.send_candidate(candidate)
File "/usr/share/gajim/src/common/jingle_content.py", line 122, in...# Bug description
Traceback (most recent call last):
File "/usr/share/gajim/src/common/jingle_rtp.py", line 178, in _on_gst_message
self.send_candidate(candidate)
File "/usr/share/gajim/src/common/jingle_content.py", line 122, in send_candidate
self.session.send_transport_info(content)
File "/usr/share/gajim/src/common/jingle_session.py", line 263, in send_transport_info
self.connection.connection.send(stanza)
File "/usr/share/gajim/src/common/xmpp/dispatcher_nb.py", line 531, in send
self._owner.Connection.send(stanza, now)
File "/usr/share/gajim/src/common/xmpp/transports_nb.py", line 490, in send
r = self.encode_stanza(raw_data)
File "/usr/share/gajim/src/common/xmpp/transports_nb.py", line 507, in encode_stanza
stanza = ustr(stanza).encode('utf-8')
File "/usr/share/gajim/src/common/xmpp/simplexml.py", line 45, in ustr
r = what.__str__()
File "/usr/share/gajim/src/common/xmpp/simplexml.py", line 169, in __str__
s = s + a.__str__(fancy and fancy+1)
File "/usr/share/gajim/src/common/xmpp/simplexml.py", line 169, in __str__
s = s + a.__str__(fancy and fancy+1)
File "/usr/share/gajim/src/common/xmpp/simplexml.py", line 169, in __str__
s = s + a.__str__(fancy and fancy+1)
File "/usr/share/gajim/src/common/xmpp/simplexml.py", line 151, in __str__
s = (fancy-1) * 2 * ' ' + "\<" + self.name
TypeError: cannot concatenate 'str' and 'Node' objects
# Steps to reproduce
Start or receive a call with/from a Pidgin 2.6.2 user
# Software versions
OS version: Ubuntu 9.10
GTK version: 2.18.3
PyGTK version: 2.16.00.14ThibgThibghttps://dev.gajim.org/gajim/gajim/-/issues/5524Bug when starting a video chat with a Pidgin 2.6.2 user2018-04-29T20:38:58ZanonymousBug when starting a video chat with a Pidgin 2.6.2 user# Bug description
Traceback (most recent call last):
File "/usr/share/gajim/src/common/jingle_rtp.py", line 178, in _on_gst_message
self.send_candidate(candidate)
File "/usr/share/gajim/src/common/jingle_content.py", line 118, in...# Bug description
Traceback (most recent call last):
File "/usr/share/gajim/src/common/jingle_rtp.py", line 178, in _on_gst_message
self.send_candidate(candidate)
File "/usr/share/gajim/src/common/jingle_content.py", line 118, in send_candidate
content = self.__content()
File "/usr/share/gajim/src/common/jingle_content.py", line 110, in __content
return xmpp.Node('content',
NameError: global name 'xmpp' is not defined
# Steps to reproduce
-> Start a video-chat with a pidgin user who have Jingle
-> The bug appears, the webcam say it's recording, but the contact don't see me !
# Software versions
OS version: Ubuntu 9.10
GTK version: 2.18.3
PyGTK version: 2.16.00.14ThibgThibghttps://dev.gajim.org/gajim/gajim/-/issues/8113Gajim to Gajim jingle on Windows doesn't work2018-04-30T11:40:50ZanonymousGajim to Gajim jingle on Windows doesn't work# Bug description
Using Gajim on Windows communicating with jingle to another Gajim on Windows does not work. I confirmed this in comparison to using jingle from Gajim Windows to Pidgin Linux which works just fine.
# Steps to reproduce
U...# Bug description
Using Gajim on Windows communicating with jingle to another Gajim on Windows does not work. I confirmed this in comparison to using jingle from Gajim Windows to Pidgin Linux which works just fine.
# Steps to reproduce
Use audio communication with the Windows version of Gajim to another contact with another user of Gajim Windows
# Software versions
OS version: 0.16.2 - Confirmed for Windows 7, Windows 8, Windows 8.1 and Windows 10
GTK version: 2.24.10
PyGTK version: 2.24.0https://dev.gajim.org/gajim/gajim/-/issues/7544Increase the key size for Jingle XTLS, enable TLS 1.1 and TLS 1.22018-04-30T11:42:45Zfedor.brunnerIncrease the key size for Jingle XTLS, enable TLS 1.1 and TLS 1.2# problem
* The Jingle XTLS implemented Gajim uses only 1024 bit RSA key. This is quite weak.
* The Jingle XTLS communication in Gajim support only TLS v1, not TLS v1.1 and TLS v1.2
* The private key ~/.config/gajim/localcert.pkey is ha...# problem
* The Jingle XTLS implemented Gajim uses only 1024 bit RSA key. This is quite weak.
* The Jingle XTLS communication in Gajim support only TLS v1, not TLS v1.1 and TLS v1.2
* The private key ~/.config/gajim/localcert.pkey is has permission 0644, it's should have only 0600
* MD5 is used as certificate signature algorithm, MD5 in X509 certificates was broken
# analysis
It's important to make Jingle XTLS stronger, in future it could be used to authenticate VoIP encryption, ZRTP (XEP-0262) or DTLS (XEP-0320).
Replace the MD5 signature algorithm with SHA-1. SHA-2 256 bit would be better but it's not supported in older OpenSSL versions.
Most web servers are currently using 2048 bit RSA keys. 3072-bit RSA keys should be equivalent in strength to 128-bit symmetric keys. My personal favorite is 4096 bit RSA, this gives enough reserve into future.
https://en.wikipedia.org/wiki/Key_size
TLS 1 and OP_SINGLE_DH_USE, the same as in
https://python-nbxmpp.gajim.org/ticket/8
# enhancement recommendation
diff -r 0861ddcb7fab src/common/jingle_xtls.py
--- a/src/common/jingle_xtls.py Wed Oct 30 11:58:18 2013 +0100
+++ b/src/common/jingle_xtls.py Sun Nov 03 23:33:57 2013 +0100
@@ -92,7 +92,10 @@
"""
constructs and returns the context objects
"""
- ctx = SSL.Context(SSL.TLSv1_METHOD)
+ ctx = SSL.Context(SSL.SSLv23_METHOD)
+ flags = (SSL.OP_NO_SSLv2 | SSL.OP_NO_SSLv3 | SSL.OP_SINGLE_DH_USE)
+ ctx.set_options(flags)
+ ctx.set_cipher_list('HIGH:!aNULL:!eNULL')
if fingerprint == 'server': # for testing purposes only
ctx.set_verify(SSL.VERIFY_NONE|SSL.VERIFY_FAIL_IF_NO_PEER_CERT,
@@ -174,12 +177,12 @@
pkey.generate_key(type, bits)
return pkey
-def createCertRequest(pkey, digest="md5", **name):
+def createCertRequest(pkey, digest="sha1", **name):
"""
Create a certificate request.
Arguments: pkey - The key to associate with the request
- digest - Digestion method to use for signing, default is md5
+ digest - Digestion method to use for signing, default is sha1
**name - The name of the subject of the request, possible
arguments are:
C - Country name
@@ -201,7 +204,7 @@
req.sign(pkey, digest)
return req
-def createCertificate(req, (issuerCert, issuerKey), serial, (notBefore, notAfter), digest="md5"):
+def createCertificate(req, (issuerCert, issuerKey), serial, (notBefore, notAfter), digest="sha1"):
"""
Generate a certificate given a certificate request.
@@ -213,7 +216,7 @@
starts being valid
notAfter - Timestamp (relative to now) when the certificate
stops being valid
- digest - Digest method to use for signing, default is md5
+ digest - Digest method to use for signing, default is sha1
Returns: The signed certificate in an X509 object
"""
cert = crypto.X509()
@@ -233,10 +236,12 @@
and '.cert' extensions
CN : common name
"""
- key = createKeyPair(TYPE_RSA, 1024)
+ key = createKeyPair(TYPE_RSA, 4096)
req = createCertRequest(key, CN=CN)
cert = createCertificate(req, (req, key), 0, (0, 60*60*24*365*5)) # five years
- open(filepath + '.pkey', 'w').write(crypto.dump_privatekey(
+ private_key_file = open(filepath + '.pkey', 'w')
+ os.chmod(filepath + '.pkey', 0600)
+ private_key_file.write(crypto.dump_privatekey(
crypto.FILETYPE_PEM, key))
open(filepath + '.cert', 'w').write(crypto.dump_certificate(
crypto.FILETYPE_PEM, cert))0.16ThibgThibghttps://dev.gajim.org/gajim/gajim/-/issues/7113Port to Farstream2018-04-30T11:42:49ZanonymousPort to Farstream# Bug description
Farsight2, currently used to provide VoIP in Gajim has been deprecated and is being replaced by Farstream. They are not parallel installable and GNOME 3.4 will depend on Farstream, so here is a patch to update Gajim to...# Bug description
Farsight2, currently used to provide VoIP in Gajim has been deprecated and is being replaced by Farstream. They are not parallel installable and GNOME 3.4 will depend on Farstream, so here is a patch to update Gajim to the new API.
This patch is against the hg trunk0.15.1ThibgThibghttps://dev.gajim.org/gajim/gajim/-/issues/6852Keep aspect ratio of video window (video calls)2018-04-30T11:42:53ZanonymousKeep aspect ratio of video window (video calls)Hello,
I have added option "Keep original aspect ratio of video window" for video calls.
It's just about one parameter for gstreamer. Most of the changes in the attached patch are related to new configuration parameter (one checkbutton...Hello,
I have added option "Keep original aspect ratio of video window" for video calls.
It's just about one parameter for gstreamer. Most of the changes in the attached patch are related to new configuration parameter (one checkbutton).
Let me know please, if there is anything you don't like.
Thanks!
Tomas0.14.2ThibgThibghttps://dev.gajim.org/gajim/gajim/-/issues/7157SRTP support2023-11-26T22:05:13ZanonymousSRTP support# problem
Gajim doesn't support Secure RTP for jingle calls.
# analysis
The SRTP support is required by WebRTC specification so the Gajim need to support it in order to be able to make calls to VoIP web applications.
Protocol descript...# problem
Gajim doesn't support Secure RTP for jingle calls.
# analysis
The SRTP support is required by WebRTC specification so the Gajim need to support it in order to be able to make calls to VoIP web applications.
Protocol description: http://xmpp.org/extensions/xep-0167.html#srtp
# enhancement recommendation
Unfortunately, I don't know any existent Jingle implementations that supports SRTP to test it (except of probably jitsi but I can't test it anyway and it prefers ZRTP but probably can serve incoming SRTP calls as well).ThibgThibghttps://dev.gajim.org/gajim/gajim/-/issues/9381Audio/Video device selection not possible2018-10-14T10:18:16ZherrwuselAudio/Video device selection not possible## Versions
- OS: Kubuntu 18.04
- Gajim version: 1.0.1
- GTK version: 3.22.30
- Python-nbxmpp version: 0.6.4
## Steps to reproduce the problem
1. Open settings and go to audio /video
2. Can't select anything although there is ...## Versions
- OS: Kubuntu 18.04
- Gajim version: 1.0.1
- GTK version: 3.22.30
- Python-nbxmpp version: 0.6.4
## Steps to reproduce the problem
1. Open settings and go to audio /video
2. Can't select anything although there is should be options
## Expected behavior
Select at least mic device and output device
## Actual behavior
see above
I only found an issue about Windows.https://dev.gajim.org/gajim/gajim/-/issues/9515Unable to start video session with Wayland2020-03-17T19:28:04ZCalle KaboUnable to start video session with Wayland## Versions
- OS: Debian GNU/Linux buster/sid
- GTK+ Version: 3.24.2
- PyGObject Version: 3.30.4
- python-nbxmpp Version: 0.6.8
- Gajim Version: 1.1.0
## Traceback
```
Traceback (most recent call last):
File "/usr/lib/python3/dist-pac...## Versions
- OS: Debian GNU/Linux buster/sid
- GTK+ Version: 3.24.2
- PyGObject Version: 3.30.4
- python-nbxmpp Version: 0.6.8
- Gajim Version: 1.1.0
## Traceback
```
Traceback (most recent call last):
File "/usr/lib/python3/dist-packages/gajim/chat_control.py", line 363, in _on_video
self.on_jingle_button_toggled(state, 'video')
File "/usr/lib/python3/dist-packages/gajim/chat_control.py", line 751, in on_jingle_button_toggled
out_xid = out_da.get_window().get_xid()
AttributeError: 'GdkWaylandWindow' object has no attribute 'get_xid'
```
## Steps to reproduce the problem
- Open a chat window
- Click the hamburger menu
- Tick the box `Video Session`1.2.0https://dev.gajim.org/gajim/gajim/-/issues/9535Jingle Audio: Update codecs (Patch)2019-07-10T20:32:13ZOliJingle Audio: Update codecs (Patch)[jingle_rtp.py.patch](/uploads/4b26b99f0e555b40b615d3d99add8c6a/jingle_rtp.py.patch)
This patch enables the Speex codec without promoting it to the preferred default codec. speex/16000 was hard coded as the preferred codec, which effect...[jingle_rtp.py.patch](/uploads/4b26b99f0e555b40b615d3d99add8c6a/jingle_rtp.py.patch)
This patch enables the Speex codec without promoting it to the preferred default codec. speex/16000 was hard coded as the preferred codec, which effectively prevented the use of the Opus codec between Gajim clients (since the introduction of Opus in farstream). speex/32000 is also added.
Codecs are offered in this order now:
opus/48000
speex/32000
speex/16000
g722/8000
speex/8000
pcma/8000
pcmu/8000
before:
speex/16000
speex/8000
opus/48000
[...]
all other codecs like L16 (16bit, 44.1khz, 1.4Mb/s), MPA (MPEG-2), MP3, or patented Siren are disabled now.
1.2.0https://dev.gajim.org/gajim/gajim/-/issues/9540Ideas / tasks for making Jingle audio calls great again2023-05-28T10:14:38ZOliIdeas / tasks for making Jingle audio calls great again### Documentation
* [ ] update https://dev.gajim.org/gajim/gajim/wikis/help/gajimfaq#general
* [ ] add wiki page about audio calls
### Codecs
* [x] ~~Add iLBC and GSM as low bandwidth fallbacks for older voip clients (?)~~
* [x] Add ~~A...### Documentation
* [ ] update https://dev.gajim.org/gajim/gajim/wikis/help/gajimfaq#general
* [ ] add wiki page about audio calls
### Codecs
* [x] ~~Add iLBC and GSM as low bandwidth fallbacks for older voip clients (?)~~
* [x] Add ~~AMR-WB and~~ AMR codecs.
* [ ] Figure out how to set the bit rate for Opus.
### DSP
* [x] Echo canceller
### UI
* [ ] Info about the used codec, bit rate and STUN
* [x] Some icons for calling / hang-up? 📞
* [ ] Preferences: setting for bit rate preferences / max bit rate. could be just a slider. codecs are disabled or reordered accordingly (don't forget rtp overhead).
* [ ] Preferences: `[x] Enable STUN auto discovery (recommended)`
### Connection
* [ ] STUN auto discovery.
### Testing
* [ ] Jitsi
* [ ] Some qxmpp client (telepathy-nonsense?)
* [ ] aTalk
* [ ] Monal
* [ ] Asterisk
* [ ] Sylkserver: Echo and Playback
* [ ] Jingle-SIP-Gateway. If that works, test SIP clients:
* [ ] Linphone
* [ ] Fritzbox
* [ ] native Android SIP client
* [ ] baresip
* [ ] PhonerLite
* [ ] MicroSIP
### Open questions
* Is it possible to use a TURN server with farstream?
* Are there any web clients that do support Jingle audio?
* Is ICE supported (XEP-0371)?https://dev.gajim.org/gajim/gajim/-/issues/9690Error in stream setup when starting audio session2019-04-29T16:49:58ZDanielError in stream setup when starting audio session## Versions
- OS: Manjaro Linux
- GTK+ Version: 3.24.8
- PyGObject Version: 3.32.0
- python-nbxmpp Version: 0.6.10
- Gajim Version: 1.1.2
## Traceback
```
Traceback (most recent call last):
File "/usr/lib/python3.7/site-packages/gajim...## Versions
- OS: Manjaro Linux
- GTK+ Version: 3.24.8
- PyGObject Version: 3.32.0
- python-nbxmpp Version: 0.6.10
- Gajim Version: 1.1.2
## Traceback
```
Traceback (most recent call last):
File "/usr/lib/python3.7/site-packages/gajim/chat_control.py", line 359, in _on_audio
self.on_jingle_button_toggled(state, 'audio')
File "/usr/lib/python3.7/site-packages/gajim/chat_control.py", line 763, in on_jingle_button_toggled
'start_' + jingle_type)(self.contact.get_full_jid())
File "/usr/lib/python3.7/site-packages/gajim/common/jingle.py", line 127, in start_audio
jingle.add_content('voice', JingleAudio(jingle))
File "/usr/lib/python3.7/site-packages/gajim/common/jingle_rtp.py", line 337, in __init__
self.setup_stream()
File "/usr/lib/python3.7/site-packages/gajim/common/jingle_rtp.py", line 352, in setup_stream
JingleRTPContent.setup_stream(self, self._on_src_pad_added)
File "/usr/lib/python3.7/site-packages/gajim/common/jingle_rtp.py", line 108, in setup_stream
self.p2pstream.set_transmitter_ht('nice', params)
gi.repository.GLib.GError: fs-error: Unknown transmitter: nice (100)
```
## Steps to reproduce the problem
Open a contact chat dialog on gajim
Click to "audio session"https://dev.gajim.org/gajim/gajim/-/issues/9757Video session doesn't work2019-07-12T14:56:32ZNikolay AmiantovVideo session doesn't work**Please first check if another issue has been opened for your problem**
## Versions
- OS: NixOS (Linux)
- Gajim version: 1.1.3
- GTK version: 3.24.8
- Python-nbxmpp version: 0.6.10
## Steps to reproduce the problem
1. Enab...**Please first check if another issue has been opened for your problem**
## Versions
- OS: NixOS (Linux)
- Gajim version: 1.1.3
- GTK version: 3.24.8
- Python-nbxmpp version: 0.6.10
## Steps to reproduce the problem
1. Enable "Video session" in a chat.
## Expected behavior
Video stream works.
## Actual behavior
1. Error in terminal:
```
x264 [error]: baseline profile doesn't support 4:2:2
```
2. Error message box (couldn't copy text):
![Screenshot_from_2019-07-12_17-50-13](/uploads/5e2c284c6ff0ec993437c3cf98dfbbc0/Screenshot_from_2019-07-12_17-50-13.png)
Maybe it's related to a specific webcam - no idea.https://dev.gajim.org/gajim/gajim/-/issues/9805gi.repository.GLib.GError: fs-error: Could not create the nicesrc element (1)2021-04-23T11:08:23ZJ. R. Schmidgi.repository.GLib.GError: fs-error: Could not create the nicesrc element (1)## Versions
- OS: Linux
- GTK+ Version: 3.24.10
- PyGObject Version: 3.32.2
- python-nbxmpp Version: 0.6.10
- Gajim Version: 1.1.2+6c3bdb4dcfb0
## Traceback
```
Traceback (most recent call last):
File "/usr/lib/python3.6/site-packages...## Versions
- OS: Linux
- GTK+ Version: 3.24.10
- PyGObject Version: 3.32.2
- python-nbxmpp Version: 0.6.10
- Gajim Version: 1.1.2+6c3bdb4dcfb0
## Traceback
```
Traceback (most recent call last):
File "/usr/lib/python3.6/site-packages/nbxmpp/dispatcher_nb.py", line 502, in dispatch
handler['func'](session, stanza)
File "/usr/lib/python3.6/site-packages/gajim/common/jingle.py", line 111, in _JingleCB
self._sessions[sid].on_stanza(stanza)
File "/usr/lib/python3.6/site-packages/gajim/common/jingle_session.py", line 354, in on_stanza
call(stanza=stanza, jingle=jingle, error=error, action=action)
File "/usr/lib/python3.6/site-packages/gajim/common/jingle_session.py", line 525, in __on_session_initiate
contents, _contents_rejected, reason_txt = self.__parse_contents(jingle)
File "/usr/lib/python3.6/site-packages/gajim/common/jingle_session.py", line 635, in __parse_contents
content = content_type(self, transport=transport)
File "/usr/lib/python3.6/site-packages/gajim/common/jingle_rtp.py", line 337, in __init__
self.setup_stream()
File "/usr/lib/python3.6/site-packages/gajim/common/jingle_rtp.py", line 352, in setup_stream
JingleRTPContent.setup_stream(self, self._on_src_pad_added)
File "/usr/lib/python3.6/site-packages/gajim/common/jingle_rtp.py", line 108, in setup_stream
self.p2pstream.set_transmitter_ht('nice', params)
gi.repository.GLib.GError: fs-error: Could not create the nicesrc element (1)
```
## Steps to reproduce the problem
...https://dev.gajim.org/gajim/gajim/-/issues/9832Cannot launch a video session using Wayland2021-10-27T18:12:35ZRaspbeguyCannot launch a video session using Wayland## Versions
- OS: Arch Linux
- GTK+ Version: 3.24.10
- PyGObject Version: 3.32.2
- python-nbxmpp Version: 0.6.10
- Gajim Version: 1.1.3
## Traceback
```
Traceback (most recent call last):
File "/usr/lib/python3.7/site-packages/gajim/c...## Versions
- OS: Arch Linux
- GTK+ Version: 3.24.10
- PyGObject Version: 3.32.2
- python-nbxmpp Version: 0.6.10
- Gajim Version: 1.1.3
## Traceback
```
Traceback (most recent call last):
File "/usr/lib/python3.7/site-packages/gajim/chat_control.py", line 367, in _on_video
self.on_jingle_button_toggled(state, 'video')
File "/usr/lib/python3.7/site-packages/gajim/chat_control.py", line 766, in on_jingle_button_toggled
out_xid = out_da.get_window().get_xid()
AttributeError: 'GdkWaylandWindow' object has no attribute 'get_xid'
```
## Steps to reproduce the problem
Try to launch a video session.https://dev.gajim.org/gajim/gajim/-/issues/9839Audio/Video: Add hint on preferences page pointing out missing dependencies2020-03-19T20:21:39ZEugene CrosserAudio/Video: Add hint on preferences page pointing out missing dependenciesLike #7626 only on fresh linux, nightly gajim.
USB camera and audio are accessible to Firefox, Skype etc.
## Versions
- OS: Ubuntu 19.10 eoan
- Gajim version: 1.1.92+890846fc3
- GTK version: 3.24.11
- Python-nbxmpp version: 0.9...Like #7626 only on fresh linux, nightly gajim.
USB camera and audio are accessible to Firefox, Skype etc.
## Versions
- OS: Ubuntu 19.10 eoan
- Gajim version: 1.1.92+890846fc3
- GTK version: 3.24.11
- Python-nbxmpp version: 0.9.92 (nightly)
## Steps to reproduce the problem
1. Open Gajim -> Preferences, select Audio/Video tab
## Expected behavior
It should be possible to select audio and video input and output devices (and then use them for jingle calls)
## Actual behavior
All device selections are greyed out and empty. Voice chat and video chat checkboxes in the "sandwich" menu of the chat windows are greyed out.
(Could it have anything to do with apparmor?)1.2.0https://dev.gajim.org/gajim/gajim/-/issues/9977Video session fails with AttributeError: 'GstXvImageSink' object has no attri...2020-05-07T18:37:46ZNarcis GarciaVideo session fails with AttributeError: 'GstXvImageSink' object has no attribute 'set_window_handle'## Versions
- OS: Debian GNU/Linux 10 (buster)
- GTK+ Version: 3.24.5
- PyGObject Version: 3.30.4
- python-nbxmpp Version: 0.6.10
- Gajim Version: 1.1.2
## Traceback
```
Traceback (most recent call last):
File "/usr/lib/pyth...## Versions
- OS: Debian GNU/Linux 10 (buster)
- GTK+ Version: 3.24.5
- PyGObject Version: 3.30.4
- python-nbxmpp Version: 0.6.10
- Gajim Version: 1.1.2
## Traceback
```
Traceback (most recent call last):
File "/usr/lib/python3/dist-packages/gajim/common/jingle_rtp.py", line 474, in _on_sync_message
imagesink.set_window_handle(self.out_xid)
AttributeError: 'GstXvImageSink' object has no attribute 'set_window_handle'
```
## Steps to reproduce the problem
1. Connect to XMPP account
2. Double-click over a contact that is using Gajim, same version and environment.
3. Open menu icon and mark "Video session"
A small video window opens and shows own camera image during 1 or 2 seconds. After this, video window closes and application shows error/bug dialog.1.2.0https://dev.gajim.org/gajim/gajim/-/issues/10109Support DTLS-SRTP (XEP-0320) for audio/video with Conversations2023-11-26T22:05:13ZGhost UserSupport DTLS-SRTP (XEP-0320) for audio/video with Conversations## Description of the new feature
https://gist.github.com/iNPUTmice/a28c438d9bbf3f4a3d4c663ffaa224d9#notes-for-developers## Description of the new feature
https://gist.github.com/iNPUTmice/a28c438d9bbf3f4a3d4c663ffaa224d9#notes-for-developershttps://dev.gajim.org/gajim/gajim/-/issues/10158A/V preview in preferences leads to segfault on Wayland2022-06-26T14:46:57ZakallabethA/V preview in preferences leads to segfault on Wayland## Versions
- OS: Ubuntu 20.04
- Gajim version: 1.2.0 (flatpak)
- GTK version:
- Python-nbxmpp version:
## Steps to reproduce the problem
1. Open `Gajim/Preferences` goto `Audio/Video` tab
1. Go to `Advanced` tab (also s...## Versions
- OS: Ubuntu 20.04
- Gajim version: 1.2.0 (flatpak)
- GTK version:
- Python-nbxmpp version:
## Steps to reproduce the problem
1. Open `Gajim/Preferences` goto `Audio/Video` tab
1. Go to `Advanced` tab (also seems to work with any other tab)
## Expected behavior
Open the newly selected tab
## Actual behavior
Crashhttps://dev.gajim.org/gajim/gajim/-/issues/10396closing chat window with call causes error2021-01-17T18:53:33ZNESC1USclosing chat window with call causes error## Versions
- OS: Ubuntu 20.04.1 LTS
- GTK Version: 3.24.20
- PyGObject Version: 3.36.0
- GLib Version : 2.64.2
- python-nbxmpp Version: 2.0.0
- Gajim Version: 1.3.0-beta2+b060fab57
I am using ubuntu 20.10 version of ejabberd's deb pack...## Versions
- OS: Ubuntu 20.04.1 LTS
- GTK Version: 3.24.20
- PyGObject Version: 3.36.0
- GLib Version : 2.64.2
- python-nbxmpp Version: 2.0.0
- Gajim Version: 1.3.0-beta2+b060fab57
I am using ubuntu 20.10 version of ejabberd's deb package as a server, calling works between conversation clients, gajim is unable to reach the conversation client or be notified about call from it.
## Traceback
```
Traceback (most recent call last):
File "/home/blabla/python3.8/site-packages/gajim/message_window.py", line 401, in _on_window_destroy
ctrl.shutdown()
File "/home/blabla/python3.8/site-packages/gajim/chat_control.py", line 1216, in shutdown
self.close_jingle_content(jingle_type)
File "/home/blabla/python3.8/site-packages/gajim/chat_control.py", line 926, in close_jingle_content
jingle.update()
File "/home/blabla/python3.8/site-packages/gajim/chat_control.py", line 729, in update_audio
self.update_actions()
File "/home/blabla/python3.8/site-packages/gajim/chat_control.py", line 296, in update_actions
win.lookup_action(
AttributeError: 'NoneType' object has no attribute 'set_enabled'
```
## Steps to reproduce the problem
to reproduce this error:
1. open a chat window and start a call to another client
2. close it and receive the error
...https://dev.gajim.org/gajim/gajim/-/issues/10430AttributeError by enabeling live video preview2021-02-27T11:59:55ZnicoAttributeError by enabeling live video preview## Versions
- OS: Arch Linux
- GTK Version: 3.24.24
- PyGObject Version: 3.38.0
- GLib Version : 2.66.2
- python-nbxmpp Version: 2.0.1
- Gajim Version: 1.3.0
## Traceback
```
Traceback (most recent call last):
File "/usr/lib/python3.9...## Versions
- OS: Arch Linux
- GTK Version: 3.24.24
- PyGObject Version: 3.38.0
- GLib Version : 2.66.2
- python-nbxmpp Version: 2.0.1
- Gajim Version: 1.3.0
## Traceback
```
Traceback (most recent call last):
File "/usr/lib/python3.9/site-packages/gajim/gtk/settings.py", line 396, in on_switch
self.set_value(value)
File "/usr/lib/python3.9/site-packages/gajim/gtk/settings.py", line 311, in set_value
self.callback(state, self.data)
File "/usr/lib/python3.9/site-packages/gajim/gtk/preferences.py", line 960, in _toggle_live_preview
preview.toggle_preview(value)
File "/usr/lib/python3.9/site-packages/gajim/gtk/video_preview.py", line 58, in toggle_preview
return self._disable_preview()
File "/usr/lib/python3.9/site-packages/gajim/gtk/video_preview.py", line 102, in _disable_preview
self._av_pipeline.remove(self._av_src)
AttributeError: 'NoneType' object has no attribute 'remove'
```
## Steps to reproduce the problem
Enable and then disable the Live Video Preview for the AV Chat in the settings menu.1.3.1https://dev.gajim.org/gajim/gajim/-/issues/10551Package gst-plugin-gtk is missing in required dependencies2021-07-24T20:01:22ZKusonekokusoneko@kusoneko.moePackage gst-plugin-gtk is missing in required dependencies## Versions
- OS: Gentoo
- Gajim version: 1.3.1_p2 & 1.3.2
- GTK version: 4.2.1
- Python-nbxmpp version: 2.0.2-r1
## Steps to reproduce the problem
1. Click on the Gajim menu button on the top left and select Preferences
2...## Versions
- OS: Gentoo
- Gajim version: 1.3.1_p2 & 1.3.2
- GTK version: 4.2.1
- Python-nbxmpp version: 2.0.2-r1
## Steps to reproduce the problem
1. Click on the Gajim menu button on the top left and select Preferences
2. Click on Audio/Video
3. Activate Live Preview
## Expected behavior
See a live preview of whatever my webcam sees.
## Actual behavior
Getting the error message "Something went wrong. Video feature disabled"
## Additional information
My webcam is detected and works with V4L2 on a variety of software, including Zoom, Google Meet, and the following command:
`mpv --no-cache --no-osc --no-input-default-bindings --profile-low-latency --input-conf=/dev/null --title=webcam $(ls /dev/video[0,2,4,6,8] | tail -n 1)`
which makes me think the issue has to be Gajim specific or located somewhere in the chain from Gajim to getting webcam footage.
Clicking on Help -> Features shows that every feature has a checkmark.
Gajim has the following USE flags on my system (+ = enabled, - = disabled):
- +crypt : End to end encryption and GPG encryption
- -geolocation : Sharing your location
- +jingle : Audio and video calls
- -python_single_target_python3_7 : Build for Python 3.7 only
- +python_single_target_python3_8 : Build for Python 3.8 only
- -python_single_target_python3_9 : Build for Python 3.9 only
- -remote : Controlling Gajim instance from command line with gajim-remote
- +rst : Generating XHTML output from RST code
- +spell : Spellchecking of composed messages
- +upnp : Ability to request your router to forward port for file transfer
- +webp : Support WebP avatars
On the Audio/Video tab, I have the following settings set:
### Server
- Use STUN server: On
### Audio
- Audio Input Device: Pulse: Default device
- Audio Output Device: Pulse: Default device
### Video
- Video Input Device: V4L2: Default device (I also tried specifying the webcam exactly with this option, and still doesn't work any better)
- Video Framerate: Default
- Video Resolution: Default
- Show My Video Stream: On
- Live Preview: On/Off (This is the problem option, turning it on gives the error, if I leave the preferences window and come back, it's turned Off again.)
## Errors logged on stdout when started from terminal
```
05/10/2021 22:53:28 (E) gajim.gui.preview Failed to obtain a working Gstreamer GTK+ sink, video support will be disabled
```
This error makes me believe the error is somewhere between Gajim and Gstreamer.
## End message
I do not know if voice calls or video calls actually work at all as I have no friends with whom to test it.
I have raised a bug request on the Gentoo package repository to have the version bumped up to 1.3.2, but I doubt a simple version bump will fix this issue entirely.
Thanks for any help,
Kusoneko.1.3.3https://dev.gajim.org/gajim/gajim/-/issues/10742AV: Add push-to-talk and push-to-mute functionality2022-09-12T08:19:00ZGlobalSilver123AV: Add push-to-talk and push-to-mute functionalityI'm a long time Discord user who's trying to switch to a FOSS, federated and privacy respecting alternative (something like XMPP, Matrix or Jami) for voice chat,
one major roadblock I've hit is the absolute lack of voice chat clients th...I'm a long time Discord user who's trying to switch to a FOSS, federated and privacy respecting alternative (something like XMPP, Matrix or Jami) for voice chat,
one major roadblock I've hit is the absolute lack of voice chat clients that support push-to-talk and push-to-mute functionality which seem like an absolutely fundamental part of any voice chat client.
# **Requested features:**
* **push-to-talk**
* **push-to-mute**
* **ability to edit which keys are used for push-to-talk and push-to-talk**https://dev.gajim.org/gajim/gajim/-/issues/10938Tried to make an audio call and got this2022-07-30T21:34:06Zunison3Tried to make an audio call and got this## Versions:
- OS: Manjaro Linux 21.2.6 (Qonos)
- GTK Version: 3.24.34
- PyGObject Version: 3.42.1
- GLib Version : 2.72.0
- python-nbxmpp Version: 3.1.0
- Gajim Version: 1.4.3
## Traceback
```
Traceback (most recent call last):
File ...## Versions:
- OS: Manjaro Linux 21.2.6 (Qonos)
- GTK Version: 3.24.34
- PyGObject Version: 3.42.1
- GLib Version : 2.72.0
- python-nbxmpp Version: 3.1.0
- Gajim Version: 1.4.3
## Traceback
```
Traceback (most recent call last):
File "/usr/lib/python3.10/site-packages/gajim/common/jingle_rtp.py", line 143, in make_bin_from_config
gst_bin = Gst.parse_bin_from_description(pipeline, True)
gi.repository.GLib.GError: gst_parse_error: no element "webrtcdsp" (1)
During handling of the above exception, another exception occurred:
Traceback (most recent call last):
File "/usr/lib/python3.10/site-packages/gajim/gtk/controls/chat.py", line 452, in _on_start_voice_call
app.call_manager.start_call(self.account, self.jid, CallType.AUDIO)
File "/usr/lib/python3.10/site-packages/gajim/common/call_manager.py", line 313, in start_call
sid = client.get_module('Jingle').start_audio(
File "/usr/lib/python3.10/site-packages/gajim/common/modules/jingle.py", line 185, in start_audio
jingle.add_content('voice', JingleAudio(jingle))
File "/usr/lib/python3.10/site-packages/gajim/common/jingle_rtp.py", line 376, in __init__
self.setup_stream()
File "/usr/lib/python3.10/site-packages/gajim/common/jingle_rtp.py", line 430, in setup_stream
self.src_bin = self.make_bin_from_config(
File "/usr/lib/python3.10/site-packages/gajim/common/jingle_rtp.py", line 149, in make_bin_from_config
raise JingleContentSetupException
gajim.common.jingle_content.JingleContentSetupException
```
## Steps to reproduce the problem
Tried to make an audio call
...1.4.4https://dev.gajim.org/gajim/gajim/-/issues/11023Voice Call clockrate mismatch for DSP2023-01-22T12:06:02ZTorVoice Call clockrate mismatch for DSP**Please first check if another issue has been opened for your problem**
## Versions
- OS: Fedora Silverblue
- Gajim version: 1.4.5 (from [flathub.org](https://flathub.org/apps/details/org.gajim.Gajim) )
- GTK version: 3.24.34
...**Please first check if another issue has been opened for your problem**
## Versions
- OS: Fedora Silverblue
- Gajim version: 1.4.5 (from [flathub.org](https://flathub.org/apps/details/org.gajim.Gajim) )
- GTK version: 3.24.34
- Python-nbxmpp version: 3.1.0
## Steps to reproduce the problem
1. Install Gajim flatpak from flathub.org
1. Sign in with xmpp account.
1. ![Screenshot_from_2022-06-30_06-26-31](/uploads/3df7b7ccecd4758549710923ab931e0d/Screenshot_from_2022-06-30_06-26-31.png)
## Expected behavior
Expectations are that the voice call initiates properly.
## Actual behavior
Currently the window pops up and it seems to try to make a connection, then immediately hangs up. The participant does receive a missed call notification. From the commandline output I see this error:
```
(W) gajim.c.storage.archive Execution time for _commit: 161 ms
(E) gajim.c.jingle_rtp gst-stream-error-quark: Echo Probe has rate 8000 , while the DSP is running at rate 48000, use a caps filter to ensure those are the same. (11)
(E) gajim.c.jingle_rtp ../ext/webrtcdsp/gstwebrtcdsp.cpp(403): gst_webrtc_dsp_analyze_reverse_stream (): /GstPipeline:pipeline0/GstBin:bin2/GstWebrtcDsp:webrtcdsp0
(E) gajim.c.jingle_rtp gst-stream-error-quark: Internal data stream error. (1)
(E) gajim.c.jingle_rtp ../libs/gst/base/gstbasesrc.c(3127): gst_base_src_loop (): /GstPipeline:pipeline0/GstBin:bin2/GstAutoAudioSrc:autoaudiosrc0/GstPulseSrc:autoaudiosrc0-actual-src-puls:
streaming stopped, reason error (-5)
```1.7.0https://dev.gajim.org/gajim/gajim/-/issues/11361STUN server error when trying to start call2023-06-08T17:07:15ZSallySTUN server error when trying to start call## Versions
- OS: Fedora 37 Workstation On Raspberry Pi 4
- Gajim version: 1.6
- GTK version: 3.24.36
- Python-nbxmpp version: 4.0.0
## Steps to reproduce the problem
Install the app through the official Fedora repo OR from f...## Versions
- OS: Fedora 37 Workstation On Raspberry Pi 4
- Gajim version: 1.6
- GTK version: 3.24.36
- Python-nbxmpp version: 4.0.0
## Steps to reproduce the problem
Install the app through the official Fedora repo OR from flathub.
## Expected behavior
As the Video and Audio icons is there to make a call, but unfortunately every time I get an error which is the same error
```
Traceback (most recent call last):
File "/usr/lib/python3.11/site-packages/gajim/gtk/chat_stack.py", line 563, in _on_action
app.call_manager.start_call(account, jid, CallType.AUDIO)
File "/usr/lib/python3.11/site-packages/gajim/common/call_manager.py", line 314, in start_call
sid = client.get_module('Jingle').start_audio(
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
File "/usr/lib/python3.11/site-packages/gajim/common/modules/jingle.py", line 157, in start_audio
jingle.add_content('voice', JingleAudio(jingle))
^^^^^^^^^^^^^^^^^^^
File "/usr/lib/python3.11/site-packages/gajim/common/jingle_rtp.py", line 386, in __init__
self.setup_stream()
File "/usr/lib/python3.11/site-packages/gajim/common/jingle_rtp.py", line 401, in setup_stream
JingleRTPContent.setup_stream(self, self._on_src_pad_added)
File "/usr/lib/python3.11/site-packages/gajim/common/jingle_rtp.py", line 122, in setup_stream
if not stun_server and self.session.connection._stun_servers:
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
AttributeError: 'Client' object has no attribute '_stun_servers'
```
And here the logs when I run Gajim from the terminal and try to make a call:
```
gajim
No translations found for en_US
Dirs searched: [PosixPath('/home/sally/.local/share'), PosixPath('/home/sally/.local/share/flatpak/exports/share'), PosixPath('/var/lib/flatpak/exports/share'), PosixPath('/usr/local/share'), PosixPath('/usr/share')]
No plugin translation path available
(org.gajim.Gajim:19684): Gtk-CRITICAL **: 01:25:21.099: gtk_widget_get_scale_factor: assertion 'GTK_IS_WIDGET (widget)' failed
(org.gajim.Gajim:19684): Gtk-CRITICAL **: 01:25:26.523: gtk_widget_get_scale_factor: assertion 'GTK_IS_WIDGET (widget)' failed
01/08/2023 01:25:27 (W) gajim.c.m.mam (sure.im) Error from sallyhaj@sure.im: feature-not-implemented - The feature is not supported yet.
01/08/2023 01:25:27 (W) gajim.c.m.bookmarks (sure.im) Error from sallyhaj@sure.im: item-not-found
(org.gajim.Gajim:19684): libsoup-ERROR **: 01:25:32.211: libsoup3 symbols detected. Using libsoup2 and libsoup3 in the same process is not supported.
Trace/breakpoint trap (core dumped)
```
I have tried multiple XMPP server instances.
Thank you.https://dev.gajim.org/gajim/gajim/-/issues/11559Error when starting call: AttributeError 'Client' object has no attribute '_s...2023-11-18T10:32:12ZDaniel BrötzmannError when starting call: AttributeError 'Client' object has no attribute '_stun_servers'# Error Details:
- Sentry event: https://dev.gajim.org/gajim/gajim/-/error_tracking/38/details
- First seen:
2023-06-08T16:42:44+00:00
- Last seen: 2023-06-08T16:42:45+00:00
- Event: 1
- Users: 0# Error Details:
- Sentry event: https://dev.gajim.org/gajim/gajim/-/error_tracking/38/details
- First seen:
2023-06-08T16:42:44+00:00
- Last seen: 2023-06-08T16:42:45+00:00
- Event: 1
- Users: 01.8.3https://dev.gajim.org/gajim/gajim/-/issues/11671call_manager: AssertionError about account2023-11-02T21:08:40ZDaniel Brötzmanncall_manager: AssertionError about account# Error Details:
- Sentry event: https://sentry.io/gajim-aec982731/gajim/issues/4596333799
- First seen:
2023-11-02T17:24:18+00:00
- Last seen: 2023-11-02T17:24:18+00:00
- Event: 1
- Users: 0# Error Details:
- Sentry event: https://sentry.io/gajim-aec982731/gajim/issues/4596333799
- First seen:
2023-11-02T17:24:18+00:00
- Last seen: 2023-11-02T17:24:18+00:00
- Event: 1
- Users: 0https://dev.gajim.org/gajim/gajim/-/issues/11746Support calling on Windows2024-01-23T20:11:33ZJohnSupport calling on WindowsHi,
I am using Windows 10 Home and downloaded the latest version of Gajim via the Microsoft Store. It installed successfully. However in each of my 1:1 chats, both `Start Voice Call...` and `Start Video Call...` options are grayed. Whe...Hi,
I am using Windows 10 Home and downloaded the latest version of Gajim via the Microsoft Store. It installed successfully. However in each of my 1:1 chats, both `Start Voice Call...` and `Start Video Call...` options are grayed. When I go to Help > Features, there is a red X beside `Audio / Video Calls` saying: `Feature not available on Windows`. Currently, only Movim supports calling on Windows. Since Gajim is a fully featured client, I would like to make calls with Gajim using Windows. Can I make this a feature request please?
Thank you