savoirfairelinux issueshttps://git.jami.net/groups/savoirfairelinux/-/issues2022-01-03T15:43:01Zhttps://git.jami.net/savoirfairelinux/jami-project/-/issues/1373A typo in Jami Privacy Policy2022-01-03T15:43:01ZA typo in Jami Privacy PolicyIn the Jami privacy policy shown on `jami.net`
A sentence reads "We use TSL/SRTP to secure connection and communications over the network."
It should be "We use **TLS**/SRTP to secure connection and communications over the network."
Than...In the Jami privacy policy shown on `jami.net`
A sentence reads "We use TSL/SRTP to secure connection and communications over the network."
It should be "We use **TLS**/SRTP to secure connection and communications over the network."
Thank you!https://git.jami.net/savoirfairelinux/jami-client-qt/-/issues/639Homogeneize features in video-conferences and 1:1 calls2023-04-04T14:38:55ZovariHomogeneize features in video-conferences and 1:1 calls(sblin):
+ Add voice-activity, multi-stream, raiseHand and other features for video-conferences in 1:1 calls
---
Original post:
Leon: Linux Mint 20.3 Cinnamon jami-client-qt deb edition from jami.net<br>
Pat: iPad jami 3.28 from app...(sblin):
+ Add voice-activity, multi-stream, raiseHand and other features for video-conferences in 1:1 calls
---
Original post:
Leon: Linux Mint 20.3 Cinnamon jami-client-qt deb edition from jami.net<br>
Pat: iPad jami 3.28 from apple store<br>
Merv: macOS jami 2.10 from apple store
Leon, Pat and Merv all delete each other and afterwards add each other as a contact. This is to make each conversion a Swarm as advised in [Taranis, a major release of Jami](https://jami.net/taranis-a-major-release-of-jami/).
1. Leon makes an audio call with Pat (no raise/lower hand button)
1. Leon adds Merv to audio call so there is a 3-way audio conversation (still no raise/lower hand button)
1. Enable video and the raise/lower hand button appears for Leon and Pat; however there is no raise/lower hand button for Merv and Merv can not see when Leon or Pat raise their hand.
Thank youSébastien BlinSébastien Blin2023-02-28https://git.jami.net/savoirfairelinux/jami-product-backlog/-/issues/41Wrong microphone used for audio and video call2022-02-03T18:25:24Zring userWrong microphone used for audio and video call- Jami Version: Live Free or Die 20190602-01 (from F-Droid)
- Android Version: LineageOS 16 (Android 9), no GApps
- Smartphone: Samsung Galaxy S5 (klte)
If I make or receive an audio or video call, I have to talk loud and close to the m...- Jami Version: Live Free or Die 20190602-01 (from F-Droid)
- Android Version: LineageOS 16 (Android 9), no GApps
- Smartphone: Samsung Galaxy S5 (klte)
If I make or receive an audio or video call, I have to talk loud and close to the microphone at the top of my smartphone. I didn't even notice there is a second microphone at that place before.
I would expect to be able to use the mike at the button of my phone.
In a normal call without Jami I can use the "normal" microphone at the button.
Another problem is, that during an video call I have to speak directly into that micro. If I'm looking at the video stream, I'm too far away from the mike so the other user can't hear me.
Settings:
- Mobile Data: enabled (but not used for the tests)
- HD-Video: enabled
- Hardware-Acceleration: enabled
- Use system contacts: disabled
- Normal phone calls via Jami: disabled
- Start Jami while booting: enabled
- Run Jami in background: enabledBackloghttps://git.jami.net/savoirfairelinux/jami-product-backlog/-/issues/40ffmpeg: add G711, G729 codecs (SIP)2022-02-03T18:28:56Zovariffmpeg: add G711, G729 codecs (SIP)Please add the following codecs to Jami:
* [G.711a](https://en.wikipedia.org/wiki/G.711)
* [G.711u](https://en.wikipedia.org/wiki/G.711)
* [G.729a](https://en.wikipedia.org/wiki/G.729)
What do you think?
Thank you
> MyNetFone uses th...Please add the following codecs to Jami:
* [G.711a](https://en.wikipedia.org/wiki/G.711)
* [G.711u](https://en.wikipedia.org/wiki/G.711)
* [G.729a](https://en.wikipedia.org/wiki/G.729)
What do you think?
Thank you
> MyNetFone uses the **G.711a** codec, it is the standard codec used in Australia and Europe. The **G.711u** codec is used within the US, and may be present in PBXs or UC platforms from US vendors. The **G.729a** codec is often used for conference calling services.<br>
> https://www.mynetfone.com.au/support/faq/question/477
> The patents for G.711, released in 1972, have expired, so it may be used without the need for a licence
> https://en.wikipedia.org/wiki/G.711#Licensing
> As of January 1, 2017, the patent terms of most licensed patents under the G.729 Consortium have expired, the remaining unexpired patents are usable on a royalty-free basis.<br>
> https://en.wikipedia.org/wiki/G.729.1
https://git.jami.net/savoirfairelinux/ring-daemon/issues/200Backloghttps://git.jami.net/savoirfairelinux/jami-product-backlog/-/issues/39Feature Request: Push To Talk Key & Open Mic Options2022-02-16T21:39:56ZtrymeoutFeature Request: Push To Talk Key & Open Mic OptionsA cool thing to add for voice and video calls would be the ability to set a push-to-talk key for desktop (Windows, Mac, Linux) and to allow an open mic option and to add settings to configure your mic. Similar to gaming VOIP apps like Te...A cool thing to add for voice and video calls would be the ability to set a push-to-talk key for desktop (Windows, Mac, Linux) and to allow an open mic option and to add settings to configure your mic. Similar to gaming VOIP apps like TeamSpeak and Mumble.
This will allow less noise during calls and will allow users to have more privacy features on their audio during a call.Backloghttps://git.jami.net/savoirfairelinux/jami-product-backlog/-/issues/38Connectivity: be more resilient if a TURN server is not available2022-02-03T18:29:22ZSébastien BlinConnectivity: be more resilient if a TURN server is not available# Reproduce steps
+ In the settings change the turn address to something that will not work but resolvable (ex: enconn.fr)
+ Try to do a call
# Expected results
+ The call should work as soon as possible with the TURN ignored
# Curre...# Reproduce steps
+ In the settings change the turn address to something that will not work but resolvable (ex: enconn.fr)
+ Try to do a call
# Expected results
+ The call should work as soon as possible with the TURN ignored
# Current result
+ The client try to connect to the TURN server for each ICE and result with a timeout of several seconds (jami-daemon~20 on linux) for each ICE negotiations making the call just unusable
# Why?
When making a call, the first step is to gather all candidates and send this message through the DHT (or direct p2p connection if available). But to gather the TURN candidate, pjsip needs to connect to it and ask for a new session. For TCP connections, the connect() will take a lot of time to timeout (depending on /proc/sys/net/ipv4/tcp_syn_retries), for UDP I didn't dig enough to fully understand what pjsip is waiting, but I think it's something related to that candidate allocation.
# Solutions
Several solutions can be created:
1. (pjsip specific) A new timer for TURN candidate creation can be created inside pjsip, to be able to ignore TURN candidates if it's taking too long. Because Jami is a real time communication app, if the allocation is taking more than 3 seconds, this means that we are taking too much time and it's not acceptable.
2. (system + pjsip specific) Manually set the connection timeout on the sockets. For TCP, we need to do a setsockopt on TCP_SYNCNT. 2 SYN retries is acceptable imho (that's about 3 seconds. First packet + 1 retry). A solution need to be created for platforms not supporting this op. For UDP as I didn't dig enough, I don't really know what's really blocking so this will need further investigation
3. (best solution imho) Support the RFC for Trickle ICE. I don't really like 1 or 2 because sometimes TURN can work and we will ignore that fact if it's taking too long. Trickle ICE will allow us to send candidates as soon as it's gathered. This means we will be able to send separately host candidates, UPnP, relays like TURN. This is clearly the solution that will take the more time to implement, but the best solution imho.
Note for 3: the drawback I see is that, because we will send candidates separately, this will generate multiple values on the DHT instead of one. Which is a bit bad.Backloghttps://git.jami.net/savoirfairelinux/jami-product-backlog/-/issues/37Android - Audio is not automatically routed to headsets when plugged2022-02-03T18:27:04ZMohamed ChibaniAndroid - Audio is not automatically routed to headsets when pluggedOn Android, if a call is started on loudspeakers, the audio is not automatically routed to the headset if it's plugged. The user has to tap on the speaker icon on the UI, to manually route the audio to the headset.On Android, if a call is started on loudspeakers, the audio is not automatically routed to the headset if it's plugged. The user has to tap on the speaker icon on the UI, to manually route the audio to the headset.Backloghttps://git.jami.net/savoirfairelinux/jami-product-backlog/-/issues/36[Windows 10] Audio issue using specifically one output (and input?) device2022-02-16T21:39:31ZElys[Windows 10] Audio issue using specifically one output (and input?) deviceI've updated all Jami apps to the latest version (Windows 10, Android).
If you make a video call between android smartphone and android tablet the audio always works fine.
However if you make a video call between pc and any android dev...I've updated all Jami apps to the latest version (Windows 10, Android).
If you make a video call between android smartphone and android tablet the audio always works fine.
However if you make a video call between pc and any android device (tablet or smartphone) the voice input from pc has 2-3 second delay before you can hear the sound on the android device. This does not affect the sound quality. (Just audio and video are out of sync)
BUT: On the other side: The sound of the android device can be heard immediately without any delay but you can't recognize the input sound - it sounds complete different and you mainly hear noise.
This always happens and there is no way I found out to fix this issue.
I tried to change all (audio) settings but it didn't help.Backloghttps://git.jami.net/savoirfairelinux/jami-product-backlog/-/issues/35UPNP - Change port allocation/mapping policy for media (RTP/RTCP ports)2022-02-16T21:40:33ZMohamed ChibaniUPNP - Change port allocation/mapping policy for media (RTP/RTCP ports)RTP and RTCP port numbers are determined using port ranges from user configuration. The ranges are configured per media type (audio and video). The current behavior has few issues such:
- The configuration is limited to 2 streams: one au...RTP and RTCP port numbers are determined using port ranges from user configuration. The ranges are configured per media type (audio and video). The current behavior has few issues such:
- The configuration is limited to 2 streams: one audio stream and one video stream.
- Pre-defined port numbers will conflict with UPNP port mappings.
The required changes are as follows:
- Remove the restriction of media stream count to 2 streams.
- Do not use pre-defined port when requesting UPNP ports.
- Configuration of media port ranges should not be necessary anymore.Backloghttps://git.jami.net/savoirfairelinux/jami-product-backlog/-/issues/34ICE/PJNATH - Use a single STUN config to add multiple SRFLX candidates2022-02-03T20:25:46ZMohamed ChibaniICE/PJNATH - Use a single STUN config to add multiple SRFLX candidatesBackloghttps://git.jami.net/savoirfairelinux/jami-product-backlog/-/issues/33Custom OpenDHT bootstrap server2022-02-03T21:48:11ZArtemy GavrelyukCustom OpenDHT bootstrap serverI haven't find an option to use a custom OpenDHT bootstrap server. Seems like it's absent like a custom TURN-server (jami-client-ios#60). It could be great to have both of them. In my case it could help to investigate huge lags in messag...I haven't find an option to use a custom OpenDHT bootstrap server. Seems like it's absent like a custom TURN-server (jami-client-ios#60). It could be great to have both of them. In my case it could help to investigate huge lags in message delivery.Backloghttps://git.jami.net/savoirfairelinux/jami-product-backlog/-/issues/32macOs/Daemon - Account can not re-register to DHT using the same port2022-02-03T20:26:22ZSébastien BlinmacOs/Daemon - Account can not re-register to DHT using the same portOn macOs, registering the account to the DHT sometimes fail with "Address already in use".
The client/system seems to fail to close the DHT socket, or does not close it fast enough.On macOs, registering the account to the DHT sometimes fail with "Address already in use".
The client/system seems to fail to close the DHT socket, or does not close it fast enough.Backloghttps://git.jami.net/savoirfairelinux/jami-product-backlog/-/issues/31UPNP/ICE/TCP - ICE/TCP connection using relay should be reset if a UPNP/NAT-P...2022-02-03T20:27:35ZMohamed ChibaniUPNP/ICE/TCP - ICE/TCP connection using relay should be reset if a UPNP/NAT-PMP is availableBackloghttps://git.jami.net/savoirfairelinux/jami-product-backlog/-/issues/30Conference: Show remote recording indication for non-host in a conference2022-02-03T20:28:46ZSébastien BlinConference: Show remote recording indication for non-host in a conference# Scenario
+ Start a conf A(host),B,C
+ B start to record
# Current result
only A,B have the remote recording indication
# Expected
C show the remote recording# Scenario
+ Start a conf A(host),B,C
+ B start to record
# Current result
only A,B have the remote recording indication
# Expected
C show the remote recordingBackloghttps://git.jami.net/savoirfairelinux/jami-product-backlog/-/issues/29Conference: Implement conference layout2022-02-16T21:41:55ZSébastien BlinConference: Implement conference layoutBackloghttps://git.jami.net/savoirfairelinux/jami-product-backlog/-/issues/28Conference: Host should be able to mute video & mic2022-02-16T21:42:15ZSébastien BlinConference: Host should be able to mute video & micBackloghttps://git.jami.net/savoirfairelinux/jami-product-backlog/-/issues/27GNU/Linux->iOS send file after call (to confirm)2022-02-16T21:43:37ZSébastien BlinGNU/Linux->iOS send file after call (to confirm)# Scenario
+ GNU/Linux call iOS
+ iOS just let ringing
+ GNU/Linux hangup
+ GNU/Linux send a file immediately after (will fail, cause iOS will cut the socket)# Scenario
+ GNU/Linux call iOS
+ iOS just let ringing
+ GNU/Linux hangup
+ GNU/Linux send a file immediately after (will fail, cause iOS will cut the socket)Backloghttps://git.jami.net/savoirfairelinux/jami-product-backlog/-/issues/26UPnP - Make port ranges configurable2022-02-03T20:29:33ZMohamed ChibaniUPnP - Make port ranges configurableThe port ranges (TCP and UDP) used by upnp to port mapping allocation must be configurable (currently they are hard-coded).
A new API must be added to allow a user to set the ranges for both UDP and TCP ports. If not set, default values ...The port ranges (TCP and UDP) used by upnp to port mapping allocation must be configurable (currently they are hard-coded).
A new API must be added to allow a user to set the ranges for both UDP and TCP ports. If not set, default values be used.
Might be related to https://git.jami.net/savoirfairelinux/ring-daemon/-/issues/417Backloghttps://git.jami.net/savoirfairelinux/jami-product-backlog/-/issues/25Recheck all APIs2022-02-03T20:34:22ZSébastien BlinRecheck all APIs+ Remove unnecessary APIs
+ Homogeneize file naming
+ definitions URI/ids
+ Split ConfigurationManager+ Remove unnecessary APIs
+ Homogeneize file naming
+ definitions URI/ids
+ Split ConfigurationManagerBackloghttps://git.jami.net/savoirfairelinux/jami-product-backlog/-/issues/24UPNP-NATPMP - Add a unit test with minimal validation2022-02-03T20:34:58ZMohamed ChibaniUPNP-NATPMP - Add a unit test with minimal validationBacklog