savoirfairelinux issueshttps://git.jami.net/groups/savoirfairelinux/-/issues2024-01-25T19:27:07Zhttps://git.jami.net/savoirfairelinux/jami-daemon/-/issues/947Blind transfer doesn't work2024-01-25T19:27:07ZSébastien BlinBlind transfer doesn't work# Scenario
+ Create SIP account
+ Do a blind transfer (transfer to another contact without an active call)
# Expected
The transfer is successful
# Current result
No transfer happen (however attended transfer works fine)# Scenario
+ Create SIP account
+ Do a blind transfer (transfer to another contact without an active call)
# Expected
The transfer is successful
# Current result
No transfer happen (however attended transfer works fine)Sébastien BlinSébastien Blinhttps://git.jami.net/savoirfairelinux/jami-client-qt/-/issues/1514SIP-Account: Rename "Delete Account" and remove the deleting warning2024-02-09T17:00:14ZElysSIP-Account: Rename "Delete Account" and remove the deleting warningusing Jami on desktop
I think we should rename "Delete account" so it will be like "Remove your SIP account" and "Remove SIP account" (You can't really delete the SIP account using Jami? It depends on your provider (server).
![SIP-dele...using Jami on desktop
I think we should rename "Delete account" so it will be like "Remove your SIP account" and "Remove SIP account" (You can't really delete the SIP account using Jami? It depends on your provider (server).
![SIP-deleteAccount](/uploads/39879462abd070e4a6bd35c478aff4e6/SIP-deleteAccount.png)
And we should either remove the warning dialog or add something like "This will remove your SIP-Account from Jami. You will need to add the SIP-account and login again"
![SIP-deleteAccount-dialog](/uploads/33b5716e27c85cfed9f4191265e2d955/SIP-deleteAccount-dialog.png)https://git.jami.net/savoirfairelinux/jami-client-android/-/issues/1364SIP account "randomly" disconnect with "connection error"2023-09-13T18:20:26ZThomas M.SIP account "randomly" disconnect with "connection error"Hello
I am new to Jami. I use it as a SIP client, with ntfy for push notifications
It appears that sometimes, “randomly”, at least I cannot reproduce it, it disconnects
It is written “Connection Error” and in the logs, I have read: “SIP ...Hello
I am new to Jami. I use it as a SIP client, with ntfy for push notifications
It appears that sometimes, “randomly”, at least I cannot reproduce it, it disconnects
It is written “Connection Error” and in the logs, I have read: “SIP registration failed, status=500 (Registering glare condition)”
I have to toggle on/off to reconnect. Sometimes it reconnects by itself after a while
What could it be? I tried different app, and two are installed at the same time (Jami and another one)
There could be a conflict?
I have already read some thread looking this one, but they date from 2021, they are maybe outdated?
Thank you in advance for your help
specifications:
* Another SIP client than Jami is installed on the smartphone
* I use e/OS and not "regular" Android
* I use a VPN
* It seems it only happens on Wifi, not on mobile data (with mobile data, I do not use the VPN)
Here is the log I exported finding the account disconnected (I hope I deleted all personal data...)
Thank you in advance for your help
<details><summary>Click to expand</summary>
```
[xxxxxxxxx.xxx|19808|manager.cpp :720 ] Not initialized
[xxxxxxxxx.xxx|19808|manager.cpp :794 ] Using PJSIP version 2.12.1 for aarch64-unknown-linux-android
[xxxxxxxxx.xxx|19808|manager.cpp :795 ] Using GnuTLS version 3.8.0
[xxxxxxxxx.xxx|19808|manager.cpp :796 ] Using OpenDHT version 2.6.0
[xxxxxxxxx.xxx|19808|manager.cpp :797 ] Using FFmpeg version 4.0.0-3315-g9899ae3f2
[xxxxxxxxx.xxx|19808|manager.cpp :800 ] Using Libgit2 version 1.6.4
[xxxxxxxxx.xxx|19808|sipvoiplink.cpp :742 ] SIPVoIPLink@xxxxxxxxxxxx
[xxxxxxxxx.xxx|19808|manager.cpp :816 ] Configuration file path: /data/user/0/cx.ring/app_config/dring.yml
[xxxxxxxxx.xxx|19808|system_codec_container.cpp:220 ] Can't find a usable accelerated H265/HEVC codec, disabling.
[xxxxxxxxx.xxx|19808|system_codec_container.cpp:250 ] Encoders found: H264 VP8 MP4V-ES H263-1998 opus G722 G726-32 speex speex speex PCMA PCMU
[xxxxxxxxx.xxx|19808|system_codec_container.cpp:251 ] Decoders found: H264 VP8 MP4V-ES H263-1998 opus G722 G726-32 speex speex speex PCMA PCMU
[xxxxxxxxx.xxx|19808|sipaccount.cpp :1476] Presence enabled for xxxxxxxxxxxxxxxx : false.
[xxxxxxxxx.xxx|19808|ringbuffer.cpp :55 ] Create new RingBuffer urgentRingBuffer_id
[xxxxxxxxx.xxx|19808|audiolayer.cpp :65 ] [audiolayer] AGC: 1, noiseReduce: auto, VAD: 1, echoCancel: auto, audioProcessor: webrtc
[xxxxxxxxx.xxx|19808|sipaccount.cpp :1476] Presence enabled for xxxxxxxxxxxxxxxx : false.
[xxxxxxxxx.xxx|19808|sipaccount.cpp :566 ] doRegister xxx.xxxx.fr
[xxxxxxxxx.xxx|19808|sipvoiplink.cpp :1521] try to resolve 'xxx.xxxx.fr' (port: 0)
[xxxxxxxxx.xxx|19808|sipaccount.cpp :663 ] Creating transport
[xxxxxxxxx.xxx|19808|siptransport.cpp :332 ] Created UDP transport on address 0.0.0.0:5060
[xxxxxxxxx.xxx|19808|siptransport.cpp :83 ] SipTransport@xxxxxxxxxxxx tr=xxxxxxxxxxxx rc=2
[xxxxxxxxx.xxx|19808|sipaccount.cpp :316 ] Set new transport [xxxxxxxxxxxx]
[xxxxxxxxx.xxx|19808|sipaccount.cpp :755 ] Using contact header <sip:xx.x.x.x2xxxx@xx.x.x.x:5060> in registration
[xxxxxxxxx.xxx|19808|sipaccount.cpp :528 ] [SIP Account xxxxxxxxxxxxxxxx] setPushNotificationToken:
[xxxxxxxxx.xxx|19824|sipaccount.cpp :1565] [Account xxxxxxxxxxxxxxxx] Checking IP route after the registration
[xxxxxxxxx.xxx|19824|sipaccount.cpp :1590] Checking received VIA address: 1xx.x.x.x.125
[xxxxxxxxx.xxx|19824|sipaccount.cpp :1671] [account xxxxxxxxxxxxxxxx] Contact address changed: (xx.x.x.x:5060 --> 1xx.x.x.x.1xx.x.x.x). Updating registration.
[xxxxxxxxx.xxx|19824|sipaccount.cpp :887 ] New contact: <sip:003xxx.xxx.xx.xxx.2.1xx.x.x.x9238>
[xxxxxxxxx.xxx|19808|configurationmanager.cpp:967 ] received connectivity changed - trying to re-connect enabled accounts
[xxxxxxxxx.xxx|19808|sipaccount.cpp :307 ] Removing old transport [xxxxxxxxxxxx] from account
[xxxxxxxxx.xxx|19808|siptransport.cpp :104 ] ~SipTransport@xxxxxxxxxxxx tr=xxxxxxxxxxxx rc=11
[xxxxxxxxx.xxx|19808|sipaccount.cpp :316 ] Set new transport [0x0]
[xxxxxxxxx.xxx|19808|sipaccount.cpp :566 ] doRegister xxx.xxxx.fr
[xxxxxxxxx.xxx|19808|sipvoiplink.cpp :1521] try to resolve 'xxx.xxxx.fr' (port: 0)
[xxxxxxxxx.xxx|19808|sipaccount.cpp :663 ] Creating transport
[xxxxxxxxx.xxx|19808|siptransport.cpp :295 ] Recycling transport 0.0.0.0:5060
[xxxxxxxxx.xxx|19808|siptransport.cpp :83 ] SipTransport@xxxxxxxxxxxx tr=xxxxxxxxxxxx rc=11
[xxxxxxxxx.xxx|19808|sipaccount.cpp :316 ] Set new transport [xxxxxxxxxxxx]
[xxxxxxxxx.xxx|19808|sipaccount.cpp :755 ] Using contact header <sip:xx.x.x.x2xxxx@xx.x.x.x:5060> in registration
[xxxxxxxxx.xxx|19825|message_engine.cpp :346 ] [Account xxxxxxxxxxxxxxxx] saved 0 messages to /data/user/0/cx.ring/cache/xxxxxxxxxxxxxxxx/messages
[xxxxxxxxx.xxx|19808|configurationmanager.cpp:967 ] received connectivity changed - trying to re-connect enabled accounts
[xxxxxxxxx.xxx|19808|sipaccount.cpp :957 ] pjsip_regc_send failed with error 171001: Object is busy (PJSIP_EBUSY)
[xxxxxxxxx.xxx|19808|sipaccount.cpp :701 ] doUnregister VoipLinkException occurred: Unable to send request to unregister sip account
[xxxxxxxxx.xxx|19808|sipaccount.cpp :307 ] Removing old transport [xxxxxxxxxxxx] from account
[xxxxxxxxx.xxx|19808|siptransport.cpp :104 ] ~SipTransport@xxxxxxxxxxxx tr=xxxxxxxxxxxx rc=15
[xxxxxxxxx.xxx|19808|sipaccount.cpp :316 ] Set new transport [0x0]
[xxxxxxxxx.xxx|19808|sipaccount.cpp :566 ] doRegister xxx.xxxx.fr
[xxxxxxxxx.xxx|19808|sipvoiplink.cpp :1521] try to resolve 'xxx.xxxx.fr' (port: 0)
[xxxxxxxxx.xxx|19808|sipaccount.cpp :663 ] Creating transport
[xxxxxxxxx.xxx|19808|siptransport.cpp :295 ] Recycling transport 0.0.0.0:5060
[xxxxxxxxx.xxx|19808|siptransport.cpp :83 ] SipTransport@xxxxxxxxxxxx tr=xxxxxxxxxxxx rc=15
[xxxxxxxxx.xxx|19808|sipaccount.cpp :316 ] Set new transport [xxxxxxxxxxxx]
[xxxxxxxxx.xxx|19808|sipaccount.cpp :755 ] Using contact header <sip:xx.x.x.x2xxxx@xx.x.x.x:5060> in registration
[xxxxxxxxx.xxx|19808|sipaccount.cpp :528 ] [SIP Account xxxxxxxxxxxxxxxx] setPushNotificationToken:
[xxxxxxxxx.xxx|19808|configurationmanager.cpp:967 ] received connectivity changed - trying to re-connect enabled accounts
[xxxxxxxxx.xxx|19808|sipaccount.cpp :957 ] pjsip_regc_send failed with error 171001: Object is busy (PJSIP_EBUSY)
[xxxxxxxxx.xxx|19808|sipaccount.cpp :701 ] doUnregister VoipLinkException occurred: Unable to send request to unregister sip account
[xxxxxxxxx.xxx|19808|sipaccount.cpp :307 ] Removing old transport [xxxxxxxxxxxx] from account
[xxxxxxxxx.xxx|19808|siptransport.cpp :104 ] ~SipTransport@xxxxxxxxxxxx tr=xxxxxxxxxxxx rc=21
[xxxxxxxxx.xxx|19808|sipaccount.cpp :316 ] Set new transport [0x0]
[xxxxxxxxx.xxx|19808|sipaccount.cpp :566 ] doRegister xxx.xxxx.fr
[xxxxxxxxx.xxx|19808|sipvoiplink.cpp :1521] try to resolve 'xxx.xxxx.fr' (port: 0)
[xxxxxxxxx.xxx|19808|sipaccount.cpp :663 ] Creating transport
[xxxxxxxxx.xxx|19808|siptransport.cpp :295 ] Recycling transport 0.0.0.0:5060
[xxxxxxxxx.xxx|19808|siptransport.cpp :83 ] SipTransport@xxxxxxxxxxxx tr=xxxxxxxxxxxx rc=21
[xxxxxxxxx.xxx|19808|sipaccount.cpp :316 ] Set new transport [xxxxxxxxxxxx]
[xxxxxxxxx.xxx|19808|sipaccount.cpp :755 ] Using contact header <sip:xx.x.x.x2xxxx@xx.x.x.x:5060> in registration
[xxxxxxxxx.xxx|19824|sipaccount.cpp :854 ] SIP registration failed, status=500 (Registering glare condition)
[xxxxxxxxx.xxx|19824|sipaccount.cpp :1773] Scheduling re-registration retry in 51 seconds..
[xxxxxxxxx.xxx|19816] Connectivity change check: host address xx.x.x.x
[xxxxxxxxx.xxx|19808|sipaccount.cpp :528 ] [SIP Account xxxxxxxxxxxxxxxx] setPushNotificationToken:
```
</details>https://git.jami.net/savoirfairelinux/jami-client-android/-/issues/1322[SIP-Account] Android version of Jami still shows "Block contact"; Deleting (...2023-07-14T13:34:52ZElys[SIP-Account] Android version of Jami still shows "Block contact"; Deleting (SIP-)contact takes longusing Jami(20230710-01) on Android
If you receive a call the caller will be added to the contact list.
- If you tap the contact you'll still get the option "Block contact" (which does NOT work https://git.jami.net/savoirfairelinux/jami...using Jami(20230710-01) on Android
If you receive a call the caller will be added to the contact list.
- If you tap the contact you'll still get the option "Block contact" (which does NOT work https://git.jami.net/savoirfairelinux/jami-client-qt/-/issues/1207)? So you should remove it
- If you try to delete a caller (Name "Anonymous" since the caller used a private number) from the contact list it can take very long (like 30 seconds or longer)https://git.jami.net/savoirfairelinux/jami-client-android/-/issues/1314SIP call with SRTP doesn't work, RTP fallback doesn't work either2023-07-25T08:31:05ZPierre NicolasSIP call with SRTP doesn't work, RTP fallback doesn't work either## Describe your environment
- Device model: Samsung SM-T720
- Android version: Android 11 API30
- What build you are using: your own android:94f816d399ba5f24af1670dfea99b87da1497a6e daemon:3c25f607f4f52028409fc0ea119ca943550e55e0...## Describe your environment
- Device model: Samsung SM-T720
- Android version: Android 11 API30
- What build you are using: your own android:94f816d399ba5f24af1670dfea99b87da1497a6e daemon:3c25f607f4f52028409fc0ea119ca943550e55e0
## Steps to reproduce
- Can you reproduce the bug: at will
- Steps:
1. Active param Encrypt media streams (SRTP)
2. Try to place or receive a cal
- Actual result: it doesn't work
3. Active param Fallback to RTP if SRTP fails
- Actual result: it doesn't fallbackhttps://git.jami.net/savoirfairelinux/jami-client-android/-/issues/1312[SIP-Account] Can't answer a SIP-call2023-07-14T18:17:25ZElys[SIP-Account] Can't answer a SIP-callusing Jami 20230616-01 on Android 13 (Samsung Galaxy A53 5G)
might be related: https://git.jami.net/savoirfairelinux/jami-client-android/-/issues/1302
Step to reproduce:
- In the past you've created a SIP-account, used it successfully ...using Jami 20230616-01 on Android 13 (Samsung Galaxy A53 5G)
might be related: https://git.jami.net/savoirfairelinux/jami-client-android/-/issues/1302
Step to reproduce:
- In the past you've created a SIP-account, used it successfully and disabled it for a few weeks (never used again; however other clients were used for the same SIP-account; but the SIP-Account in Jami has been always disabled; so the phone has been probably rebooted many times)
- (Now you disable all other clients for the same SIP-account) Now enable the SIP-account in Jami again
- Make a call (Phone 1 (SIM) -> Phone 2(SIP-Account; using Jami) )
- You'll get a notification "Decline" or "Answer in audio"
- If you tap "Answer in audio" nothing will happen but another noficiation "Answer in audio" will appear
- You can tap like 4 next notification "Answer in audio" but nothing will happen
- After that the call will fail
- You'll get a "missed call" notification
NOTE:
- Enabeling/Disabeling the STUN-option didn't change anything
- Enabeling/Disabeling the SIP-account didn't change anything
- Switching between wifi and mobile networks didn't change anythingPierre NicolasPierre Nicolashttps://git.jami.net/savoirfairelinux/jami-client-android/-/issues/1302SIP Account: Number shown twice2023-07-13T16:56:12ZElysSIP Account: Number shown twiceusing the newest beta version of Android
Steps to reproduce?:
- Add a SIP-Account
- You'll view your number
- But you get the same account twice? (You can only disable/enable one account; the other will be automatically enabled / disab...using the newest beta version of Android
Steps to reproduce?:
- Add a SIP-Account
- You'll view your number
- But you get the same account twice? (You can only disable/enable one account; the other will be automatically enabled / disabled too)
![SIP-issue](/uploads/bd45559eccb0be3c9c9f4907aa6bef5c/SIP-issue.png)https://git.jami.net/savoirfairelinux/jami-client-qt/-/issues/1173[SIP] Delete account and then Jami disappears (closes or quits)2023-07-31T12:53:10Zovari[SIP] Delete account and then Jami disappears (closes or quits)1. Have one Jami account and one SIP account
2. Delete the SIP account, then Jami disappears (closes or quits) and the Jami icon in the system tray also disappears
Are you able to reproduce and please fix?
Thank you
Linux Mint 21.1 Ci...1. Have one Jami account and one SIP account
2. Delete the SIP account, then Jami disappears (closes or quits) and the Jami icon in the system tray also disappears
Are you able to reproduce and please fix?
Thank you
Linux Mint 21.1 Cinnamon<br>
Jami Version: 202305271132
P.S. Unable to verify once fixed as no longer have any SIP account. Hopefully the fix will be beneficial to Jami and JAMS users who have SIP accounts.Aline Gondim SantosAline Gondim Santoshttps://git.jami.net/savoirfairelinux/jami-client-android/-/issues/1253Impossible to establish ip2ip SIP call (at least android->desktop and desktop...2023-04-24T15:22:18ZPierre NicolasImpossible to establish ip2ip SIP call (at least android->desktop and desktop->android)## Describe your environment
- Device model: Samsung Galaxy Tab S5e SM-T720
- Android version: 11
## Steps to reproduce
- Can you reproduce the bug: at will
- Steps:
1. Create and connect SIP account on Android (ip2ip UDP)
1...## Describe your environment
- Device model: Samsung Galaxy Tab S5e SM-T720
- Android version: 11
## Steps to reproduce
- Can you reproduce the bug: at will
- Steps:
1. Create and connect SIP account on Android (ip2ip UDP)
1. Create and connect SIP account on Desktop (ip2ip UDP)
1. Connect your devices on same network
2. Try to make a ip2ip voice call from android, accept from desktop
- Actual result: Call connecting activity disappear without establishing call. Missed outgoing call appears on conversation. It continues ringing. Have to restart the app.
## Additional information
Looking through logs above make think it come from failing media negotiation.
Call logs from desktop to android :
```
[1682347249.158|40881|channeled_transport.cpp :186 ] [SIPS] process disconnect event
[1682347249.158|40880|ice_transport.cpp :1694] [ice:0x7fb0b8cd5570] ice send failed: Not found (PJ_ENOTFOUND)
[1682347249.158|40881|siptransport.cpp :209 ] pjsip transport@0x7fb184219c90 TLS to 142.170.109.216 -> DISCONNECTED
[1682347249.158|56389|siptransport.cpp :100 ] ~SipTransport@0x7faf08032e90 tr=0x7faf0809bfb0 rc=1
[1682347249.158|40880|tls_session.cpp :893 ] [TLS] transport failure on tx: errno = 5
[1682347249.159|56389|gitserver.cpp :477 ] GitServer destroyed
[1682347249.159|56389|siptransport.cpp :100 ] ~SipTransport@0x7fb184216bc0 tr=0x7fb184219c90 rc=1
[1682347249.159|56815|ice_transport.cpp :336 ] [ice:0x7fb0b8cd5570] destroying 0x7fb0b8dee168
[1682347249.659|56815|ice_transport.cpp :350 ] [ice:0x7fb0b8cd5570] Destroying ice_strans 0x7fb0b8dee168
[1682347250.159|56815|ice_transport.cpp :669 ] [ice:0x7fb0b8cd5570] Timer heap flushed after 500ms
[1682347250.159|56815|ice_transport.cpp :382 ] [ice:0x7fb0b8cd5570] done destroying
[1682347250.985|56392|sipvoiplink.cpp :892 ] [call:6027552833534573] INVITE@0x7fb0b9fe44c8 state changed to 4 (CONNECTING): cause=0, tsx@0x7fb14482de48 status 200 (OK)
[1682347250.985|56392|sipvoiplink.cpp :1121] [call:6027552833534573] INVITE@0x7fb0b9fe44c8 media update: status 220048
[1682347250.985|56392|sipvoiplink.cpp :1129] [call:6027552833534573] SDP offer failed, reason 415
[1682347250.985|56392|sipcall.cpp :756 ] [call:6027552833534573] Terminate SIP session
```
Call logs from desktop to android :
```
[1682347307.380|56356|manager.cpp :1047] Answer call 2214803505268786
[1682347307.380|56356|audiostream.cpp :162 ] Destroying stream with device alsa_output.pci-0000_0a_00.3.iec958-stereo
[1682347307.380|56356|sipcall.cpp :889 ] [call:2214803505268786] Answering incoming call with following media:
[1682347307.380|56356|sipcall.cpp :892 ] [call:2214803505268786] Media @0 - type [AUDIO] enabled [YES] muted [NO] label [audio_0] source [camera://046d_0821_0C411BC0] secure [NO]
[1682347307.380|56356|sipcall.cpp :2333] [call:2214803505268786] [audio_0] already un-muted
[1682347307.380|56356|sdp.cpp :604 ] Processing received offer for [Call ID 2214803505268786] with 1 media
[1682347307.380|56356|sdp.cpp :503 ] [SDP OFFER] Remote session:
v=0
o=localhost 3891336105 0 IN IP4 192.168.0.217
s=Call ID 7629516086156536
c=IN IP4 192.168.0.217
t=0 0
a=ice-ufrag:4825cdac
a=ice-pwd:3c5a176c6406842d7f98f7c5
m=audio 18770 RTP/SAVP 104 9 2 112 111 110 8 0 101
a=rtpmap:104 opus/48000/2
a=rtpmap:9 G722/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:112 speex/32000
a=rtpmap:111 speex/16000
a=rtpmap:110 speex/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtcp:18771 IN IP4 192.168.0.217
a=sendrecv
a=candidate:Hc0a800d9 1 UDP 2130706431 192.168.0.217 48819 typ host
a=candidate:Ha556a7c0 1 UDP 2130706431 fe80::2c47:5fff:fe35:d569 51892 typ host
a=candidate:Hc0a800d9 2 UDP 2130706430 192.168.0.217 41158 typ host
a=candidate:Ha556a7c0 2 UDP 2130706430 fe80::2c47:5fff:fe35:d569 36779 typ host
[1682347307.380|56356|sdp.cpp :263 ] Add media description [type [AUDIO] enabled [YES] muted [NO] label [audio_0] source [camera://046d_0821_0C411BC0] secure [NO]]
[1682347307.381|56356|sdp.cpp :503 ] [SDP ANSWER] Local session:
v=0
o=atlas 3891336107 0 IN IP4 192.168.49.92
s=Call ID 2214803505268786
c=IN IP4 192.168.49.92
t=0 0
m=audio 26904 RTP/AVP 104 9 2 112 111 110 8 0 101
a=rtpmap:104 opus/48000/2
a=rtpmap:9 G722/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:112 speex/32000
a=rtpmap:111 speex/16000
a=rtpmap:110 speex/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtcp:26905 IN IP4 192.168.49.92
a=sendrecv
[1682347307.381|56356|sipcall.cpp :3498] [call:2214803505268786] Setup ICE response
[1682347307.382|56356|ice_transport.cpp :331 ] [ice:0x25216e0] Creating IceTransport session for "2214803505268786"
[1682347307.382|56356|sipcall.cpp :3312] [call:2214803505268786] Successfully created media ICE transport [ice:0x5fe8c10]
[1682347307.382|56356|sipcall.cpp :3474] [call:2214803505268786] Setting ICE session [0x5fe8c10]
[1682347307.382|56356|sipcall.cpp :3334] [call:2214803505268786] Init media ICE transport
[1682347307.382|56356|ice_transport.cpp :406 ] [ice:0x25216e0] Initializing the session - comp count 2 - as a slave
[1682347307.382|56356|ice_transport.cpp :447 ] [ice:0x25216e0] Add host candidates
[1682347307.382|56356|ice_transport.cpp :906 ] [ice:0x25216e0] added host stun config for UDP transport
[1682347307.382|56356|ice_transport.cpp :906 ] [ice:0x25216e0] added host stun config for UDP transport
[1682347307.382|56356|ice_transport.cpp :906 ] [ice:0x25216e0] added host stun config for UDP transport
[1682347307.382|56356|ice_transport.cpp :989 ] [ice:0x25216e0] Add srflx reflexive candidates [192.168.49.92:21799 : 192.168.49.92:21799] for comp 1
[1682347307.382|56356|ice_transport.cpp :989 ] [ice:0x25216e0] Add srflx reflexive candidates [192.168.49.92:22898 : 192.168.49.92:22898] for comp 2
[1682347307.382|56356|ice_transport.cpp :469 ] [ice:0x25216e0] Added generic srflx candidates:
[1682347307.387|56356|ice_transport.cpp :707 ] [ice:0x25216e0] UDP initialization success
[1682347307.387|56356|ice_transport.cpp :787 ] [ice:0x25216e0] as slave
[1682347307.388|56356|ice_transport.cpp :881 ] [ice:0x25216e0] (local) ufrag=5d1495c7, pwd=29e8d5446da38dae04605688
[1682347307.388|56356|sipcall.cpp :1872] [call:2214803505268786] Add local attributes for ICE instance [0x5fe8c10]
[1682347307.388|56356|sipcall.cpp :1912] [call:2214803505268786] add ICE local candidates for media [type [AUDIO] enabled [YES] muted [NO] label [audio_0]] @ 0
[1682347307.388|56356|sipvoiplink.cpp :1121] [call:2214803505268786] INVITE@0x7fb0b8b329e8 media update: status 0
[1682347307.388|56356|sdp.cpp :139 ] Set active local session to [0x52c4148]. Was [(nil)]
[1682347307.388|56356|sdp.cpp :503 ] [SDP ANSWER] Local active session:
v=0
o=atlas 3891336107 1 IN IP4 192.168.49.92
s=Call ID 2214803505268786
c=IN IP4 192.168.49.92
t=0 0
a=ice-ufrag:5d1495c7
a=ice-pwd:29e8d5446da38dae04605688
m=audio 0 RTP/SAVP 104 9 2 112 111 110 8 0 101
[1682347307.388|56356|sdp.cpp :147 ] Set active remote session to [0x245f5a8]. Was [(nil)]
[1682347307.388|56356|sdp.cpp :503 ] [SDP ANSWER] Remote active session:
v=0
o=localhost 3891336105 0 IN IP4 192.168.0.217
s=Call ID 7629516086156536
c=IN IP4 192.168.0.217
t=0 0
a=ice-ufrag:4825cdac
a=ice-pwd:3c5a176c6406842d7f98f7c5
m=audio 18770 RTP/SAVP 104 9 2 112 111 110 8 0 101
a=rtpmap:104 opus/48000/2
a=rtpmap:9 G722/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:112 speex/32000
a=rtpmap:111 speex/16000
a=rtpmap:110 speex/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtcp:18771 IN IP4 192.168.0.217
a=sendrecv
a=candidate:Hc0a800d9 1 UDP 2130706431 192.168.0.217 48819 typ host
a=candidate:Ha556a7c0 1 UDP 2130706431 fe80::2c47:5fff:fe35:d569 51892 typ host
a=candidate:Hc0a800d9 2 UDP 2130706430 192.168.0.217 41158 typ host
a=candidate:Ha556a7c0 2 UDP 2130706430 fe80::2c47:5fff:fe35:d569 36779 typ host
[1682347307.388|56356|sipcall.cpp :968 ] [call:2214803505268786] Answering with contact header: <sip:192.168.49.92:5062>
[1682347307.388|56356|sipvoiplink.cpp :892 ] [call:2214803505268786] INVITE@0x7fb0b8b329e8 state changed to 4 (CONNECTING): cause=0, tsx@0x7fb0b404aa98 status 200 (OK)
[1682347307.388|56356|call.cpp :241 ] [call:2214803505268786] state change 0/1, cnx 3/4, code 0
[1682347307.388|56356|call.cpp :275 ] [call:2214803505268786] emit client call state change CURRENT, code 0
[1682347307.389|56356|manager.cpp :603 ] ----- Switch current call id to '2214803505268786' -----
[1682347307.389|56389|sipcall.cpp :2605] [call:2214803505268786] Media negotiation complete
[1682347307.389|56356|manager.cpp :1614] Add audio to call 2214803505268786
[1682347307.389|56389|sipcall.cpp :2671] [call:2214803505268786] Starting ICE
[1682347307.389|56356|manager.cpp :1625] [call:2214803505268786] Attach audio
[1682347307.389|56356|ringbufferpool.cpp :174 ] Bind call 2214803505268786 to call audiolayer_id
[1682347307.389|56389|sdp.cpp :941 ] Media#0 is disabled. Media ports: local 0, remote 18770
[1682347307.389|56356|ringbufferpool.cpp :155 ] Bind rbuf '2214803505268786' to callid 'audiolayer_id'
[1682347307.389|56389|ice_transport.cpp :1231] [ice:0x25216e0] start failed: no remote candidates[1682347307.389|56356|ringbufferpool.cpp :155 ] Bind rbuf 'audiolayer_id' to callid '2214803505268786'
[1682347307.389|56389|sipcall.cpp :2702] [call:2214803505268786] ICE media failed to start
[1682347307.389|56356|audiostream.cpp :50 ] Playback: Creating stream with device (48000Hz, 2 chan
```https://git.jami.net/savoirfairelinux/jami-client-qt/-/issues/1073CallOverlay: incorrect button shown sometimes2023-06-13T12:43:10ZSébastien BlinCallOverlay: incorrect button shown sometimes# Scenario
+ Do a call with a SIP account, hangup
+ Do a call with a Jami Account, hangup
+ Do a call with a SIP account
# Expected
+ Pause button, call forward, DTMF panel MUST be shown
# Current result
+ Sometimes pause/forward/dt...# Scenario
+ Do a call with a SIP account, hangup
+ Do a call with a Jami Account, hangup
+ Do a call with a SIP account
# Expected
+ Pause button, call forward, DTMF panel MUST be shown
# Current result
+ Sometimes pause/forward/dtmf are not shown
# Note
CallActionBar.qml is the root cause. This file seems badly designed depending on a lot of reset() instead of just showing properties correctly. I'd recommend to re-do this classAline Gondim SantosAline Gondim Santoshttps://git.jami.net/savoirfairelinux/jami-client-android/-/issues/1239SIP call on android: plain RTP session impossible (intentionally unencrypted ...2023-04-14T12:29:07ZTobias HuberSIP call on android: plain RTP session impossible (intentionally unencrypted LAN-PBX)Hi, since Android 13 doesn't offer native SIP support anymore, I thought this great Jami thing could step in...
Unfortunately, it was quiet inconvenient to get at least some idea why this doesn't work with my LAN/VPN PBX
_(Inconvenience...Hi, since Android 13 doesn't offer native SIP support anymore, I thought this great Jami thing could step in...
Unfortunately, it was quiet inconvenient to get at least some idea why this doesn't work with my LAN/VPN PBX
_(Inconvenience: LAN/VPN PBX is asterisk 1.6, which doesn't use identical string for account identification and user authentication; the user corresponding to the password for authentication is different to the name of the SIP account.
It's possible to set Jami up that way, but it's quiet hidden - besides some more not so minor UI nits on android, but that will be a different issue report.)_
I checked that all Security->Security switches are off (SRTP and TLS transport).
1.) Initiating (outgoing) call fails because:
`"Rejecting secure audio stream without encryption details: audio 19686 RTP/SAVP 104 9 2 112 111 110 8 0 101"`
> <--- SIP read from UDP:172.21.97.226:5060 ---> [115/1865]
> INVITE sip:11@pbx.example.net SIP/2.0
> Via: SIP/2.0/UDP 172.21.97.226:5060;rport;branch=z9hG4bKPjaf1c522c-6e92-4539-b899-15c2bad55ad1
> Max-Forwards: 70
> From: <sip:tobimob_line1@pbx.example.net>;tag=b2110c22-6e14-4381-8a99-cc1e40d3872f
> To: <sip:11@pbx.example.net>
> Contact: <sip:tobimob_line1@172.21.97.226:5060>
> Call-ID: 4b970369-1b13-44da-b798-3a74f51d8c39
> CSeq: 629 INVITE
> Subject: Phone call
> Allow: SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, OPTIONS, MESSAGE, PUBLISH
> Supported: replaces
> User-Agent: Jami Daemon 13.7.0 (android)
> Authorization: Digest username="tobi.huber", realm="pbx.example.net", nonce="3b54e054", uri="sip:11@pbx.example.net", response="4070f25aadda2536ee24058551dfb64
> 1", algorithm=MD5
> Content-Type: application/sdp
> Content-Length: 1047
>
> v=0
> o=localhost 3890394360 0 IN IP4 172.21.97.226
> s=Call ID 8706183892147327
> c=IN IP4 172.21.97.226
> t=0 0
> a=ice-ufrag:42a09c4e
> a=ice-pwd:04afb39763f13c7f32a67b91
> m=aud--- (15 headers 27 lines) ---
> Sending to 172.21.97.226:5060 (no NAT)
> Using INVITE request as basis request - 4b970369-1b13-44da-b798-3a74f51d8c39
> Found peer 'tobimob_line1' for 'tobimob_line1' from 172.21.97.226:5060
> Found RTP audio format 104
> Found RTP audio format 9
> Found RTP audio format 2
> Found RTP audio format 112
> Found RTP audio format 111
> Found RTP audio format 110
> Found RTP audio format 8
> Found RTP audio format 0
> Found RTP audio format 101
> Found unknown media description format opus for ID 104
> Found audio description format G722 for ID 9
> Found audio description format G726-32 for ID 2
> Found audio description format speex for ID 112
> Found audio description format speex for ID 111
> Found audio description format speex for ID 110
> Found audio description format PCMA for ID 8
> Found audio description format PCMU for ID 0
> Found audio description format telephone-event for ID 101
[Apr 13 19:05:59] ERROR[101987][C-000005d5]: chan_sip.c:33575 int setup_srtp(struct sip_srtp **): No SRTP module loaded, can't setup SRTP session.
[Apr 13 19:05:59] WARNING[101987][C-000005d5]: chan_sip.c:10417 int process_sdp(struct sip_pvt *, struct sip_request *, int): Rejecting secure audio stream witho
ut encryption details: audio 19686 RTP/SAVP 104 9 2 112 111 110 8 0 101
Why does it send RTP/SAVP although I disabled SRTP?
2.) Incoming call signalling works, but not possible to establish RTP session:
```
"Ignoring audio media offer because port number is zero" and
"Failing due to no acceptable offer found"
```
Ringing:
> <--- SIP read from UDP:172.21.97.226:5060 --->
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP 172.21.98.12:5060;received=172.21.98.12;branch=z9hG4bK0d2bf1b6
> Call-ID: 1e4bf56a62264ba6792450e4506bed52@pbx.example.net
> From: "T. Huber (22)" <sip:thbuero@pbx.example.net>;tag=as48fb14ca
> To: <sip:tobimob_line1@172.21.97.226>;tag=6c456d21-e6ae-4c85-bf04-da0aaf56e758
> CSeq: 102 INVITE
> User-Agent: Jami Daemon 13.7.0 (android)
> Allow: SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, OPTIONS, MESSAGE, PUBLISH
> Contact: <sip:tobimob_line1@172.21.97.226:5060>
... (snipped)
Accepting call on Jami:
> <--- SIP read from UDP:172.21.97.226:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 172.21.98.12:5060;received=172.21.98.12;branch=z9hG4bK0d2bf1b6
> Call-ID: 1e4bf56a62264ba6792450e4506bed52@pbx.example.net
> From: "T. Huber (22)" <sip:thbuero@pbx.example.net>;tag=as48fb14ca
> To: <sip:tobimob_line1@172.21.97.226>;tag=6c456d21-e6ae-4c85-bf04-da0aaf56e758
> CSeq: 102 INVITE
> User-Agent: Jami Daemon 13.7.0 (android)
> Allow: SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, OPTIONS, MESSAGE, PUBLISH
> Supported: replaces
> Contact: <sip:tobimob_line1@172.21.97.226:5060>
> Content-Type: application/sdp
> Content-Length: 142
>
> v=0
> o=localhost 3890395686 1 IN IP4 172.21.97.226
> s=Call ID 5890041505306313
> c=IN IP4 172.21.97.226
> t=0 0
> m=audio 0 RTP/AVP 9 8 0 3 101
> <------------->
chan_sip.c:10008 int process_sdp(struct sip_pvt *, struct sip_request *, int): Ignoring audio media offer [5/1992]
port number is zero
chan_sip.c:10438 int process_sdp(struct sip_pvt *, struct sip_request *, int): Failing due to no acceptabl[3/1992]
found
**Here's a working internal call, where sipdroid is the user agent on the same phone:**
> <--- SIP read from UDP:172.21.97.226:40739 --->
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP 172.21.98.12:5060;branch=z9hG4bK04f5274c
> To: <sip:tobimob_line1@10.26.229.169:40739;transport=udp>;tag=fb8b4ff3c7a73ee8
> From: "T. Huber (22)" <sip:thbuero@pbx.example.net>;tag=as686012bd
> Call-ID: 3547fb4e68213eb90d8295ae221e36ea@pbx.example.net
> CSeq: 102 INVITE
> Server: Sipdroid/6.3 beta/Pixel 7
> Content-Length: 198
> Content-Type: application/sdp
... (snipped)
Accepting call on SIPdroid:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 172.21.98.12:5060;branch=z9hG4bK04f5274c
> To: <sip:tobimob_line1@10.26.229.169:40739;transport=udp>;tag=fb8b4ff3c7a73ee8
> From: "T. Huber (22)" <sip:thbuero@pbx.example.net>;tag=as686012bd
> Call-ID: 3547fb4e68213eb90d8295ae221e36ea@pbx.example.net
> CSeq: 102 INVITE
> Contact: <sip:tobimob_line1@10.26.229.169:40739;transport=udp>
> Server: Sipdroid/6.3 beta/Pixel 7
> Content-Length: 198
> Content-Type: application/sdp
>
> v=0
> o=tobimob_line1@pbx.example.net 0 0 IN IP4 10.26.229.169
> s=Session SIP/SDP
> c=IN IP4 10.26.229.169
> t=0 0
> m=audio 21000 RTP/AVP 9 101
> a=rtpmap:9 G722/8000
> a=rtpmap:101 telephone-event/8000
> <------------->
**Significant difference here is "m=audio ...." lines.**
With SIPdroid (the working incomming call), it reads
> m=audio 21000 RTP/AVP 9 101
while with jami (not working)
> m=audio 0 RTP/AVP 9 8 0 3 101
I'd very much appreciate if somebody could take care and bring back originating strenghts of Jami in that it's usable as a working SIP UA - for plain RTP too!
Especially due to dropped native SIP support in recent android versions!
The android settings UI is broken too, like already mentioned, but ther's the workaround to use it in landscape mode, which makes it possible to sroll the account-enabler switch out of overlapping tab selection area... Will tell in a different issue report.
Thanks in advance,
-Tobihttps://git.jami.net/savoirfairelinux/jami-daemon/-/issues/846SIP call on android: plain RTP session impossible (intentionally unencrypted ...2023-04-14T12:23:34ZTobias HuberSIP call on android: plain RTP session impossible (intentionally unencrypted LAN-PBX)Hi, since Android 13 doesn't offer native SIP support anymore, I thought this great Jami thing could step in...
Unfortunately, it was quiet inconvenient to get at least some idea why this doesn't work with my LAN/VPN PBX
_(Inconvenience...Hi, since Android 13 doesn't offer native SIP support anymore, I thought this great Jami thing could step in...
Unfortunately, it was quiet inconvenient to get at least some idea why this doesn't work with my LAN/VPN PBX
_(Inconvenience: LAN/VPN PBX is asterisk 1.6, which doesn't use identical string for account identification and user authentication; the user corresponding to the password for authentication is different to the name of the SIP account.
It's possible to set Jami up that way, but it's quiet hidden - besides some more not so minor UI nits on android, but that will be a different issue report.)_
I checked that all Security->Security switches are off (SRTP and TLS transport).
1.) Initiating (outgoing) call fails because:
`"Rejecting secure audio stream without encryption details: audio 19686 RTP/SAVP 104 9 2 112 111 110 8 0 101"`
> <--- SIP read from UDP:172.21.97.226:5060 ---> [115/1865]
> INVITE sip:11@pbx.example.net SIP/2.0
> Via: SIP/2.0/UDP 172.21.97.226:5060;rport;branch=z9hG4bKPjaf1c522c-6e92-4539-b899-15c2bad55ad1
> Max-Forwards: 70
> From: <sip:tobimob_line1@pbx.example.net>;tag=b2110c22-6e14-4381-8a99-cc1e40d3872f
> To: <sip:11@pbx.example.net>
> Contact: <sip:tobimob_line1@172.21.97.226:5060>
> Call-ID: 4b970369-1b13-44da-b798-3a74f51d8c39
> CSeq: 629 INVITE
> Subject: Phone call
> Allow: SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, OPTIONS, MESSAGE, PUBLISH
> Supported: replaces
> User-Agent: Jami Daemon 13.7.0 (android)
> Authorization: Digest username="tobi.huber", realm="pbx.example.net", nonce="3b54e054", uri="sip:11@pbx.example.net", response="4070f25aadda2536ee24058551dfb64
> 1", algorithm=MD5
> Content-Type: application/sdp
> Content-Length: 1047
>
> v=0
> o=localhost 3890394360 0 IN IP4 172.21.97.226
> s=Call ID 8706183892147327
> c=IN IP4 172.21.97.226
> t=0 0
> a=ice-ufrag:42a09c4e
> a=ice-pwd:04afb39763f13c7f32a67b91
> m=aud--- (15 headers 27 lines) ---
> Sending to 172.21.97.226:5060 (no NAT)
> Using INVITE request as basis request - 4b970369-1b13-44da-b798-3a74f51d8c39
> Found peer 'tobimob_line1' for 'tobimob_line1' from 172.21.97.226:5060
> Found RTP audio format 104
> Found RTP audio format 9
> Found RTP audio format 2
> Found RTP audio format 112
> Found RTP audio format 111
> Found RTP audio format 110
> Found RTP audio format 8
> Found RTP audio format 0
> Found RTP audio format 101
> Found unknown media description format opus for ID 104
> Found audio description format G722 for ID 9
> Found audio description format G726-32 for ID 2
> Found audio description format speex for ID 112
> Found audio description format speex for ID 111
> Found audio description format speex for ID 110
> Found audio description format PCMA for ID 8
> Found audio description format PCMU for ID 0
> Found audio description format telephone-event for ID 101
[Apr 13 19:05:59] ERROR[101987][C-000005d5]: chan_sip.c:33575 int setup_srtp(struct sip_srtp **): No SRTP module loaded, can't setup SRTP session.
[Apr 13 19:05:59] WARNING[101987][C-000005d5]: chan_sip.c:10417 int process_sdp(struct sip_pvt *, struct sip_request *, int): Rejecting secure audio stream witho
ut encryption details: audio 19686 RTP/SAVP 104 9 2 112 111 110 8 0 101
Why does it send RTP/SAVP although I disabled SRTP?
2.) Incoming call signalling works, but not possible to establish RTP session:
```
"Ignoring audio media offer because port number is zero" and
"Failing due to no acceptable offer found"
```
Ringing:
> <--- SIP read from UDP:172.21.97.226:5060 --->
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP 172.21.98.12:5060;received=172.21.98.12;branch=z9hG4bK0d2bf1b6
> Call-ID: 1e4bf56a62264ba6792450e4506bed52@pbx.example.net
> From: "T. Huber (22)" <sip:thbuero@pbx.example.net>;tag=as48fb14ca
> To: <sip:tobimob_line1@172.21.97.226>;tag=6c456d21-e6ae-4c85-bf04-da0aaf56e758
> CSeq: 102 INVITE
> User-Agent: Jami Daemon 13.7.0 (android)
> Allow: SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, OPTIONS, MESSAGE, PUBLISH
> Contact: <sip:tobimob_line1@172.21.97.226:5060>
... (snipped)
Accepting call on Jami:
> <--- SIP read from UDP:172.21.97.226:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 172.21.98.12:5060;received=172.21.98.12;branch=z9hG4bK0d2bf1b6
> Call-ID: 1e4bf56a62264ba6792450e4506bed52@pbx.example.net
> From: "T. Huber (22)" <sip:thbuero@pbx.example.net>;tag=as48fb14ca
> To: <sip:tobimob_line1@172.21.97.226>;tag=6c456d21-e6ae-4c85-bf04-da0aaf56e758
> CSeq: 102 INVITE
> User-Agent: Jami Daemon 13.7.0 (android)
> Allow: SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, OPTIONS, MESSAGE, PUBLISH
> Supported: replaces
> Contact: <sip:tobimob_line1@172.21.97.226:5060>
> Content-Type: application/sdp
> Content-Length: 142
>
> v=0
> o=localhost 3890395686 1 IN IP4 172.21.97.226
> s=Call ID 5890041505306313
> c=IN IP4 172.21.97.226
> t=0 0
> m=audio 0 RTP/AVP 9 8 0 3 101
> <------------->
chan_sip.c:10008 int process_sdp(struct sip_pvt *, struct sip_request *, int): Ignoring audio media offer [5/1992]
port number is zero
chan_sip.c:10438 int process_sdp(struct sip_pvt *, struct sip_request *, int): Failing due to no acceptabl[3/1992]
found
**Here's a working internal call, where sipdroid is the user agent on the same phone:**
> <--- SIP read from UDP:172.21.97.226:40739 --->
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP 172.21.98.12:5060;branch=z9hG4bK04f5274c
> To: <sip:tobimob_line1@10.26.229.169:40739;transport=udp>;tag=fb8b4ff3c7a73ee8
> From: "T. Huber (22)" <sip:thbuero@pbx.example.net>;tag=as686012bd
> Call-ID: 3547fb4e68213eb90d8295ae221e36ea@pbx.example.net
> CSeq: 102 INVITE
> Server: Sipdroid/6.3 beta/Pixel 7
> Content-Length: 198
> Content-Type: application/sdp
... (snipped)
Accepting call on SIPdroid:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 172.21.98.12:5060;branch=z9hG4bK04f5274c
> To: <sip:tobimob_line1@10.26.229.169:40739;transport=udp>;tag=fb8b4ff3c7a73ee8
> From: "T. Huber (22)" <sip:thbuero@pbx.example.net>;tag=as686012bd
> Call-ID: 3547fb4e68213eb90d8295ae221e36ea@pbx.example.net
> CSeq: 102 INVITE
> Contact: <sip:tobimob_line1@10.26.229.169:40739;transport=udp>
> Server: Sipdroid/6.3 beta/Pixel 7
> Content-Length: 198
> Content-Type: application/sdp
>
> v=0
> o=tobimob_line1@pbx.example.net 0 0 IN IP4 10.26.229.169
> s=Session SIP/SDP
> c=IN IP4 10.26.229.169
> t=0 0
> m=audio 21000 RTP/AVP 9 101
> a=rtpmap:9 G722/8000
> a=rtpmap:101 telephone-event/8000
> <------------->
**Significant difference here is "m=audio ...." lines.**
With SIPdroid (the working incomming call), it reads
> m=audio 21000 RTP/AVP 9 101
while with jami (not working)
> m=audio 0 RTP/AVP 9 8 0 3 101
I'd very much appreciate if somebody could take care and bring back originating strenghts of Jami in that it's usable as a working SIP UA - for plain RTP too!
Especially due to dropped native SIP support in recent android versions!
The android settings UI is broken too, like already mentioned, but ther's the workaround to use it in landscape mode, which makes it possible to sroll the account-enabler switch out of overlapping tab selection area... Will tell in a different issue report.
Thanks in advance,
-Tobihttps://git.jami.net/savoirfairelinux/jami-client-android/-/issues/1208New SIP conversation only appears at restart2023-07-12T15:53:35ZPierre NicolasNew SIP conversation only appears at restart## Describe your environment
- Device model: Samsung Galaxy Tab S5e SM-T720
- Android version: 11
- What build you are using: your own, commit 5be1de17c790bddfb30dde74c8a1364e213f9847
## Steps to reproduce
- Can you reproduce ...## Describe your environment
- Device model: Samsung Galaxy Tab S5e SM-T720
- Android version: 11
- What build you are using: your own, commit 5be1de17c790bddfb30dde74c8a1364e213f9847
## Steps to reproduce
- Can you reproduce the bug: at will
- Steps:
1. Open a SIP account
2. Search for a new contact and start talking to him
- Actual result: The conversation is created but your new contact doesn't appears in contact list. Furthermore, it appears at restart.
## Additional information
![output](/uploads/f7a1c29ebc9648bf453acd95c7ecbbe6/output.mp4)https://git.jami.net/savoirfairelinux/jami-client-android/-/issues/1207SIP settings are partially broken2024-01-30T14:53:27ZPierre NicolasSIP settings are partially broken## Describe your environment
- Device model: Samsung Galaxy Tab S5e SM-T720
- Android version: 11
- What build you are using: your own, commit 5be1de17c790bddfb30dde74c8a1364e213f9847
## Steps to reproduce
- Can you reproduce ...## Describe your environment
- Device model: Samsung Galaxy Tab S5e SM-T720
- Android version: 11
- What build you are using: your own, commit 5be1de17c790bddfb30dde74c8a1364e213f9847
## Steps to reproduce
- Can you reproduce the bug: at will
- Steps:
1. Open a SIP account
2. Go to settings ("Account Settings")
3. Scroll horizontally
- Actual result: Tab headers are unclickable; settings are displayed in front of headers
## Additional information
![sip_settings_broken](/uploads/cfc122de3fa6faa6ee298c7748c21cee/sip_settings_broken.mp4)Pierre NicolasPierre Nicolashttps://git.jami.net/savoirfairelinux/jami-client-qt/-/issues/1043SIP: save conversation's color in overriden vcard2023-03-22T17:22:29ZSébastien BlinSIP: save conversation's color in overriden vcard=> cf ContactModel::updateDetails
We can save conversation's preferences in it, as there is no swarm (it will not be synced)=> cf ContactModel::updateDetails
We can save conversation's preferences in it, as there is no swarm (it will not be synced)https://git.jami.net/savoirfairelinux/jami-daemon/-/issues/826problem connecting tio SIP account on startup2023-11-12T14:17:59Ztomo90problem connecting tio SIP account on startupI have configured a simple SIP account. The application does not automatically log in to the account when it starts. However, all I need to do to connect is go into the settings and click in the name settings field, for example, and then...I have configured a simple SIP account. The application does not automatically log in to the account when it starts. However, all I need to do to connect is go into the settings and click in the name settings field, for example, and then exit the settings and the application will connect to the account immediately. It doesn't make any sense to me.
Generally speaking, it will connect to the account after editing any unrelated settings.
Furthermore, the application does not seem to save some of the settings i make. After quitting, the switches and settings are at their original values. This is not a problem with, for example, write permissions to the configuration file, because some of the settings made are preserved and some are not when the application is restarted.
You can see for yourself that on the attached video.
![vid](/uploads/e0e18d538065a0cffa977629dbfcf594/vid.mp4)Sébastien BlinSébastien Blinhttps://git.jami.net/savoirfairelinux/jami-client-ios/-/issues/242SIP: adding a call doesn't add any history2023-04-18T18:17:01ZSébastien BlinSIP: adding a call doesn't add any history# Scenario
+ Create a SIP account (e.g. internal 69X)
+ Call someone (e.g. a phone)
# Expected
+ Conversation should be shown with a "Ongoing call" message# Scenario
+ Create a SIP account (e.g. internal 69X)
+ Call someone (e.g. a phone)
# Expected
+ Conversation should be shown with a "Ongoing call" messageBinal AhiyaBinal Ahiyahttps://git.jami.net/savoirfairelinux/jami-daemon/-/issues/790SIP ip 2 ip issues2022-12-08T19:39:33ZSébastien BlinSIP ip 2 ip issues+ One account on 5060, one on 5061 => 5061 receives all calls even if :5060 is specified in the URI
+ ~~text messages doesn't work~~
+ IPv6 addresses not supported
+ Calling back doesn't work+ One account on 5060, one on 5061 => 5061 receives all calls even if :5060 is specified in the URI
+ ~~text messages doesn't work~~
+ IPv6 addresses not supported
+ Calling back doesn't workhttps://git.jami.net/savoirfairelinux/jami-client-qt/-/issues/809[SIP] unable to call a contact in conversation list2022-08-23T08:11:18Zovari[SIP] unable to call a contact in conversation list## Scenario 1
1. Type a number to call in SIP account<br>
![image](/uploads/25c6f87f5b7b229b0003ac514a5301e8/image.png)
1. Call the number
2. Buttons to call that contact are no longer available<br>
![image](/uploads/e6979f8b74157359665...## Scenario 1
1. Type a number to call in SIP account<br>
![image](/uploads/25c6f87f5b7b229b0003ac514a5301e8/image.png)
1. Call the number
2. Buttons to call that contact are no longer available<br>
![image](/uploads/e6979f8b7415735966539b23f7f23edb/image.png)
## Scenario 2
1. Type a number to call in SIP account
2. Click `Add to conversations` button (located in top right)
3. Call button disappears
Are you able to reproduce and please fix?
Thank you
Linux Mint 21<br>
Jami Version: 202208152318Sébastien BlinSébastien Blinhttps://git.jami.net/savoirfairelinux/jami-client-android/-/issues/1111SIP account can't receive incoming calls with TLS activated2022-10-26T15:13:31ZLem MingSIP account can't receive incoming calls with TLS activatedWhen TLS is enabled in the SIP account, no incoming calls are received.
Environment:
- OS: Android 11
- Jami version: Taranis - 20220615-01 (F-Droid)
- Device: Teracube 2e
- SIP provider: voip.ms
Steps to reproduce:
1) Create a SIP ac...When TLS is enabled in the SIP account, no incoming calls are received.
Environment:
- OS: Android 11
- Jami version: Taranis - 20220615-01 (F-Droid)
- Device: Teracube 2e
- SIP provider: voip.ms
Steps to reproduce:
1) Create a SIP account following this [post](https://forum.jami.net/t/how-can-i-create-a-sip-account-in-the-android-version-of-jami/1294)
2) Follow the [Jami wiki on voip.ms](https://wiki.voip.ms/article/Jami) for the basic configuration
3) Activate: Encrypt media streams (SRTP)
4) Activate: Enable SDES as key exchange protocol
5) Activate: Use TLS Transport
6) Permutate between 5061 (voip.ms TLS port) and the default value for whatever port options available
Expected and actual behaviors:
- On Jami, the account is online
- Echo test work (voip.ms: 4443)
- On the portal, it is registered with secure transport
Expected behavior:
- I can receive incoming calls
Actual behavior:
- I can't receive incoming calls
Following steps:
7) Deactivate "Use TLS transport"
Actual behaviors:
- On Jami, the account is online
- Echo test work (voip.ms: 4443)
- On the portal, it is registered but **no with secure transport**
- I **can** receive incoming calls
Following steps:
8) Install Linphone for Android
9) Follow the Linphone for Android tutorial specific for voip.ms [here](https://wiki.voip.ms/article/Linphone_Android)
Expected and actual behaviors:
- On Linphone, the account is online
- Echo test work (voip.ms: 4443)
- On the portal, it is registered with secure transport
- I can receive incoming calls
For the same environment:
- same network
- same voip.ms portal configuration
- same device
it works for Linphone but not for Jami.Sébastien BlinAntoine NoreauSébastien Blin