savoirfairelinux issueshttps://git.jami.net/groups/savoirfairelinux/-/issues2023-07-25T08:31:05Zhttps://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/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/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-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/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-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-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-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-daemon/-/issues/694ffmpeg: add G711, G729 codecs (SIP)2022-07-06T00:18:18Zovariffmpeg: 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-project/-/issues/1289Bundle same contacts with different spelling2021-08-19T19:54:53ZmokkinBundle same contacts with different spellingThe following examples shows three different spellings of the same contact:
![image](/uploads/564ff72a0adad7f06a3055ba20dd53f6/image.png)
1. International format with country code and city code
2. National format with city code
3. Local...The following examples shows three different spellings of the same contact:
![image](/uploads/564ff72a0adad7f06a3055ba20dd53f6/image.png)
1. International format with country code and city code
2. National format with city code
3. Local format without any prefix
All of them are successful for calling, because the pbx/sip knows its region. For a better overview these contacts should be recognized as the same and bundled.https://git.jami.net/savoirfairelinux/jami-client-qt/-/issues/377Bugfix - SIP authentication username option missing in desktop clients2022-07-04T18:39:37ZRobinBugfix - SIP authentication username option missing in desktop clientsI'd like to refer to my request on [Twitter](https://twitter.com/jami_social/status/1217490812847763456)... BTW thanks for the quick reply :thumbsup:
---
I'm unable to set up my SIP account since there is no option for `authentication...I'd like to refer to my request on [Twitter](https://twitter.com/jami_social/status/1217490812847763456)... BTW thanks for the quick reply :thumbsup:
---
I'm unable to set up my SIP account since there is no option for `authentication username`. I have something like this:
```config
username: 1234
registrar: example.net
authentication username: johndoe
password: ***
```
Could you please fix this?Ming Rui ZhangMing Rui Zhanghttps://git.jami.net/savoirfairelinux/jami-project/-/issues/1215Please detect status 488 from SIP server and offer suggestions to change codec2023-05-29T13:37:07ZreubenfirminPlease detect status 488 from SIP server and offer suggestions to change codecWhen a SIP server returns 488, it can mean a codec incompatibility. Your UI buries that 488 is returned. Please detect the status and pop up the settings asking the user to check the codecs vs what the voip provider allows.When a SIP server returns 488, it can mean a codec incompatibility. Your UI buries that 488 is returned. Please detect the status and pop up the settings asking the user to check the codecs vs what the voip provider allows.https://git.jami.net/savoirfairelinux/jami-project/-/issues/1126[Debian 10 sid] SIP Incall on FritzBox2021-06-08T11:08:26ZPeter Maier[Debian 10 sid] SIP Incall on FritzBoxIncalls internal from FritzBox with IP phone are not working. Outcall on internal IP phones is working.
**Problem**
Add call to conversation with "**611"
"slotCallStateChanged (call: 8738021371634479), from Incoming to Talking"
(jami...Incalls internal from FritzBox with IP phone are not working. Outcall on internal IP phones is working.
**Problem**
Add call to conversation with "**611"
"slotCallStateChanged (call: 8738021371634479), from Incoming to Talking"
(jami-gnome:180595): Gtk-WARNING **: 12:57:54.298: Theme parsing error: <data>:1:79: Not using units is deprecated. Assuming 'px'.
(jami-gnome:180595): Gtk-WARNING **: 12:57:54.381: Calling org.xfce.Session.Manager.Inhibit failed: GDBus.Error:org.freedesktop.DBus.Error.UnknownMethod: No such method “Inhibit”https://git.jami.net/savoirfairelinux/jami-project/-/issues/1057No early media after SIP 1832020-10-02T21:14:46ZPaweł BogusławskiNo early media after SIP 183Hello,
When using
* jami_20200829.1.d8786b1~dfsg1-1_amd64.deb
* jami-daemon_20200829.1.d8786b1~dfsg1-1_amd64.deb
from
https://dl.jami.net/nightly/debian_10/pool/main/j/jami/
in Debian 10, Jami does not play ringback when calling ext...Hello,
When using
* jami_20200829.1.d8786b1~dfsg1-1_amd64.deb
* jami-daemon_20200829.1.d8786b1~dfsg1-1_amd64.deb
from
https://dl.jami.net/nightly/debian_10/pool/main/j/jami/
in Debian 10, Jami does not play ringback when calling external number (which uses SIP 183 message when waiting for call to be picked up and early media/SDP); when call is picked up - voice is played correctly; dring debug log of such call:
```
[1600086622.792|15229|manager.cpp :581 ] ----- Switch current call id to '2159493006188118' -----
[1600086622.792|15230|sipcall.cpp :963 ] [call:2159493006188118] fill SDP with ICE transport 0x5633ef823740
[1600086622.792|15230|sipaccount.cpp :375 ] contact header: "test" <sip:login@192.168.1.100:5060> / "test" <sip:login@sip.mydomain.loc> -> <sip:111222333@sip.mydomain.loc>
[1600086622.792|15230|sipvoiplink.cpp :853 ] [call:2159493006188118] INVITE@0x7fd6dc010a48 state changed to 1 (CALLING): cause=0, tsx@0x7fd6dc048e88 status 0 (Default status message)
[1600086622.793|15230|call.cpp :259 ] [call:2159493006188118] state change 0/1, cnx 0/2, code 0
[1600086622.793|15230|call.cpp :286 ] [call:2159493006188118] emit client call state change CONNECTING, code 0
[1600086624.793|15230|call.cpp :112 ] Call 2159493006188118 is still connecting after timeout, sending fallback request
[1600086625.482|15233|sipvoiplink.cpp :853 ] [call:2159493006188118] INVITE@0x7fd6dc010a48 state changed to 3 (EARLY): cause=0, tsx@0x7fd70800d278 status 183 (Session Progress)
[1600086625.482|15233|sipvoiplink.cpp :1009 ] [call:2159493006188118] INVITE@0x7fd6dc010a48 media update: status 0
[1600086625.482|15233|sdp.cpp :383 ] Local active SDP Session:
v=0
o=ibnb003 3809075422 0 IN IP4 192.168.1.100
s=Jami Daemon
c=IN IP4 192.168.1.100
t=0 0
a=ice-ufrag:1d26a479
a=ice-pwd:33f8ac4760b4c6501dd61dfa
m=audio 41884 RTP/AVP 110 101
a=rtpmap:110 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtcp:41885 IN IP4 192.168.1.100
a=sendrecv
a=candidate:H44e08c2a 1 UDP 2130706431 [...] 42537 typ host
a=candidate:Ha030271 1 UDP 2130706431 192.168.1.100 39418 typ host
a=candidate:Hc0a87b64 1 UDP 2130706175 [...] 39418 typ host
a=candidate:Hc0a87a01 1 UDP 2130705919 [...] 39418 typ host
a=candidate:H44e08c2a 2 UDP 2130706430 [...] 42180 typ host
a=candidate:Ha030271 2 UDP 2130706430 [...] 35281 typ host
a=candidate:Hc0a87b64 2 UDP 2130706174 [...] 35281 typ host
a=candidate:Hc0a87a01 2 UDP 2130705918 [...] 35281 typ host
m=video 0 RTP/AVP 96 97 98 99
[1600086625.482|15233|sdp.cpp :383 ] Remote active SDP Session:
v=0
o=ib 1858813877 1858813877 IN IP4 192.168.100.2
s=IB VoIP Server
c=IN IP4 192.168.100.2
t=0 0
m=audio 23562 RTP/AVP 110 101
a=rtpmap:110 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=maxptime:60
a=sendrecv
m=video 0 RTP/AVP 96 97 98 99
[1600086625.482|15233|sipcall.cpp :1230 ] [call:2159493006188118] medias changed
[1600086625.482|15230|sipcall.cpp :1242 ] [call:2159493006188118] no remote ICE for medias
[1600086625.482|15230|sipcall.cpp :1182 ] [call:2159493006188118] stopping all medias
[1600086625.482|15230|sipcall.cpp :1021 ] [call:2159493006188118] startAllMedia()
[1600086625.482|15230|socket_pair.cpp :172 ] use local address: 0.0.0.0:41884
[1600086625.482|15230|socket_pair.cpp :172 ] use local address: 0.0.0.0:41885
[1600086625.482|15230|socket_pair.cpp :349 ] SocketPair: local{41884,41885} / 192.168.100.2{23562,23563}
[1600086625.482|15230|audio_input.cpp :53 ] Creating audio input with id: 2159493006188118
[1600086625.483|15230|audio_input.cpp :270 ] Switching audio source to match 'camera://04f2b3ec200901010001'
[1600086625.483|15230|audio_sender.cpp :72 ] audioEncoder_->openOutput rtp://192.168.100.2:23562
[1600086625.483|15230|media_encoder.cpp :268 ] Not using hardware encoding for speex
[1600086625.483|15230|media_encoder.cpp :574 ] [libspeex] Using 4 threads
[1600086625.483|15230|media_encoder.cpp :621 ] [libspeex] Frame size 160
[1600086625.483|15230|sipcall.cpp :1065 ] [call:2159493006188118] [SDP:slot#1] Missing local codec
[1600086625.483|16375|media_decoder.cpp :134 ] Trying to open device dummyFilename with format sdp, pixel format , size 0x0, rate 0.000000
[1600086625.483|16375|media_decoder.cpp :146 ] Using format sdp
[1600086625.503|16374|audio_input.cpp :76 ] Switching audio input to '04f2b3ec200901010001'
[1600086628.306|15233|sipvoiplink.cpp :853 ] [call:2159493006188118] INVITE@0x7fd6dc010a48 state changed to 4 (CONNECTING): cause=0, tsx@0x7fd70800d278 status 200 (OK)
[1600086628.306|15233|sipvoiplink.cpp :1009 ] [call:2159493006188118] INVITE@0x7fd6dc010a48 media update: status 0
[1600086628.306|15233|sdp.cpp :383 ] Local active SDP Session:
v=0
o=ibnb003 3809075422 1 IN IP4 192.168.1.100
s=Jami Daemon
c=IN IP4 192.168.1.100
t=0 0
a=ice-ufrag:1d26a479
a=ice-pwd:33f8ac4760b4c6501dd61dfa
m=audio 41884 RTP/AVP 110 101
a=rtpmap:110 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtcp:41885 IN IP4 192.168.1.100
a=sendrecv
a=candidate:H44e08c2a 1 UDP 2130706431 [...] 42537 typ host
a=candidate:Ha030271 1 UDP 2130706431 [...] 39418 typ host
a=candidate:Hc0a87b64 1 UDP 2130706175 [...] 39418 typ host
a=candidate:Hc0a87a01 1 UDP 2130705919 [...] 39418 typ host
a=candidate:H44e08c2a 2 UDP 2130706430 [...] 42180 typ host
a=candidate:Ha030271 2 UDP 2130706430 [...] 35281 typ host
a=candidate:Hc0a87b64 2 UDP 2130706174 [...] 35281 typ host
a=candidate:Hc0a87a01 2 UDP 2130705918 [...] 35281 typ host
m=video 0 RTP/AVP 96 97 98 99
[1600086628.306|15233|sdp.cpp :383 ] Remote active SDP Session:
v=0
o=ib 1858813877 1858813877 IN IP4 192.168.100.2
s=IB VoIP Server
c=IN IP4 192.168.100.2
t=0 0
m=audio 23562 RTP/AVP 110 101
a=rtpmap:110 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=maxptime:60
a=sendrecv
m=video 0 RTP/AVP 96 97 98 99
[1600086628.306|15233|sipcall.cpp :1230 ] [call:2159493006188118] medias changed
[1600086628.306|15230|sipcall.cpp :1242 ] [call:2159493006188118] no remote ICE for medias
[1600086628.306|15230|sipcall.cpp :1182 ] [call:2159493006188118] stopping all medias
[1600086628.306|15233|sipvoiplink.cpp :861 ] [call:2159493006188118] INVITE@0x7fd6dc010a48 state changed to 5 (CONFIRMED): cause=0 (TX_MSG)
[1600086628.306|15233|sipcall.cpp :910 ] [call:2159493006188118] onAnswered()
[sdp @ 0x7fd6bc0206c0] Could not find codec parameters for stream 0 (Audio: speex (libspeex), 8000 Hz, mono): unspecified sample format
Consider increasing the value for the 'analyzeduration' and 'probesize' options
[1600086628.387|16375|media_decoder.cpp :481 ] Decoding audio using libspeex Speex (libspeex)
[1600086628.387|16375|media_decoder.cpp :491 ] Not using hardware decoding for speex
[1600086628.387|15230|sipcall.cpp :1021 ] [call:2159493006188118] startAllMedia()
[1600086628.387|15230|socket_pair.cpp :172 ] use local address: 0.0.0.0:41884
[1600086628.387|15230|socket_pair.cpp :172 ] use local address: 0.0.0.0:41885
[1600086628.387|15230|socket_pair.cpp :349 ] SocketPair: local{41884,41885} / 192.168.100.2{23562,23563}
[1600086628.387|15230|audio_input.cpp :270 ] Switching audio source to match 'camera://04f2b3ec200901010001'
[1600086628.387|15230|audio_sender.cpp :72 ] audioEncoder_->openOutput rtp://192.168.100.2:23562
[1600086628.387|15230|media_encoder.cpp :268 ] Not using hardware encoding for speex
[1600086628.387|15230|media_encoder.cpp :574 ] [libspeex] Using 4 threads
[1600086628.387|15230|media_encoder.cpp :621 ] [libspeex] Frame size 160
[1600086628.387|15230|sipcall.cpp :1065 ] [call:2159493006188118] [SDP:slot#1] Missing local codec
[1600086628.387|16376|media_decoder.cpp :134 ] Trying to open device dummyFilename with format sdp, pixel format , size 0x0, rate 0.000000
[1600086628.387|15230|call.cpp :259 ] [call:2159493006188118] state change 1/1, cnx 2/4, code 0
[1600086628.387|15230|call.cpp :286 ] [call:2159493006188118] emit client call state change CURRENT, code 0
[1600086628.387|16376|media_decoder.cpp :146 ] Using format sdp
[1600086628.387|15230|manager.cpp :2007 ] [call:2159493006188118] Peer answered
[1600086628.387|15230|manager.cpp :1631 ] Add audio to call 2159493006188118
[1600086628.387|15230|manager.cpp :1645 ] [call:2159493006188118] Attach audio
[1600086628.387|15230|ringbufferpool.cpp:175 ] Bind call 2159493006188118 to call audiolayer_id
[1600086628.387|15230|ringbufferpool.cpp:156 ] Bind rbuf '2159493006188118' to callid 'audiolayer_id'
[1600086628.387|15230|ringbufferpool.cpp:156 ] Bind rbuf 'audiolayer_id' to callid '2159493006188118'
[1600086628.387|15230|audiolayer.cpp :130 ] Hardware audio format available : {s16, 2 channels, 44100Hz} 0
[1600086628.387|15230|audiostream.cpp :53 ] Playback: trying to create stream with device alsa_output.usb-GN_Netcom_A_S_Jabra_PRO_930_027608F15805-00.mono-fallback (44100Hz, 1 channels)
[1600086628.387|15230|audiostream.cpp :148 ] Stream is creating...
[1600086628.387|15230|audiostream.cpp :53 ] Ringtone: trying to create stream with device (44100Hz, 2 channels)
[1600086628.387|15230|audiostream.cpp :148 ] Stream is creating...
[1600086628.387|15230|audiostream.cpp :53 ] Capture: trying to create stream with device alsa_input.usb-GN_Netcom_A_S_Jabra_PRO_930_027608F15805-00.mono-fallback (44100Hz, 1 channels)
[1600086628.387|15230|audiostream.cpp :148 ] Stream is creating...
[1600086628.403|16374|audio_input.cpp :76 ] Switching audio input to '04f2b3ec200901010001'
[1600086629.465|15245|audiostream.cpp :156 ] Stream successfully created, connected to alsa_output.usb-GN_Netcom_A_S_Jabra_PRO_930_027608F15805-00.mono-fallback
[1600086629.465|15245|audiostream.cpp :156 ] Stream successfully created, connected to alsa_output.pci-0000_00_1b.0.analog-stereo
[1600086629.465|15245|audiostream.cpp :156 ] Stream successfully created, connected to alsa_input.usb-GN_Netcom_A_S_Jabra_PRO_930_027608F15805-00.mono-fallback
[1600086629.465|15245|pulselayer.cpp :371 ] All streams ready, starting !
[1600086629.588|16376|media_decoder.cpp :481 ] Decoding audio using libspeex Speex (libspeex)
[1600086629.588|16376|media_decoder.cpp :491 ] Not using hardware decoding for speex
[1600086634.112|15233|sipvoiplink.cpp :853 ] [call:2159493006188118] INVITE@0x7fd6dc010a48 state changed to 6 (DISCONNCTD): cause=200, tsx@0x7fd70800d278 status 200 (OK)
[1600086634.112|15230|manager.cpp :2040 ] [call:2159493006188118] Peer hungup
```
Problem does not occur on same Debian system and SIP account when using Twinkle SIP client.
Problem does not occur in Jami when local VoiP number is called (which uses SIP 180 message when waiting for call, without early media/SDP); dring debug log of such /not answered/ call when ringback was played correctly:
```
[1600085689.195|15229|manager.cpp :581 ] ----- Switch current call id to '5190575654160253' -----
[1600085689.195|15230|sipcall.cpp :963 ] [call:5190575654160253] fill SDP with ICE transport 0x5633ef824e50
[1600085689.195|15230|sdp.cpp :718 ] addIceCandidates failed: cannot access media#1 (may be deactivated)
[1600085689.196|15230|sdp.cpp :718 ] addIceCandidates failed: cannot access media#1 (may be deactivated)
[1600085689.196|15230|sipaccount.cpp :375 ] contact header: "test" <sip:login@129.168.1.100:5060> / "IB" <sip:login@sip.mydomain.loc> -> <sip:login2@sip.mydomain.loc>
[1600085689.196|15230|sipvoiplink.cpp :853 ] [call:5190575654160253] INVITE@0x7fd6dc0460b8 state changed to 1 (CALLING): cause=0, tsx@0x7fd6dc0286b8 status 0 (Default status message)
[1600085689.196|15230|call.cpp :259 ] [call:5190575654160253] state change 0/1, cnx 0/2, code 0
[1600085689.196|15230|call.cpp :286 ] [call:5190575654160253] emit client call state change CONNECTING, code 0
[1600085689.374|15233|sipvoiplink.cpp :853 ] [call:5190575654160253] INVITE@0x7fd6dc0460b8 state changed to 3 (EARLY): cause=0, tsx@0x7fd70800d278 status 180 (Ringing)
[1600085689.374|15233|call.cpp :259 ] [call:5190575654160253] state change 1/1, cnx 2/3, code 0
[1600085689.374|15233|call.cpp :286 ] [call:5190575654160253] emit client call state change RINGING, code 0
[1600085689.374|15230|manager.cpp :2029 ] [call:5190575654160253] Peer ringing
[1600085689.374|15230|audiolayer.cpp :130 ] Hardware audio format available : {s16, 2 channels, 44100Hz} 0
[1600085689.374|15230|audiostream.cpp :53 ] Playback: trying to create stream with device alsa_output.usb-GN_Netcom_A_S_Jabra_PRO_930_027608F15805-00.mono-fallback (44100Hz, 1 channels)
[1600085689.374|15230|audiostream.cpp :148 ] Stream is creating...
[1600085689.374|15230|audiostream.cpp :53 ] Ringtone: trying to create stream with device (44100Hz, 2 channels)
[1600085689.374|15230|audiostream.cpp :148 ] Stream is creating...
[1600085689.374|15230|audiostream.cpp :53 ] Capture: trying to create stream with device alsa_input.usb-GN_Netcom_A_S_Jabra_PRO_930_027608F15805-00.mono-fallback (44100Hz, 1 channels)
[1600085689.374|15230|audiostream.cpp :148 ] Stream is creating...
[1600085690.454|15245|audiostream.cpp :156 ] Stream successfully created, connected to alsa_output.usb-GN_Netcom_A_S_Jabra_PRO_930_027608F15805-00.mono-fallback
[1600085690.454|15245|audiostream.cpp :156 ] Stream successfully created, connected to alsa_output.pci-0000_00_1b.0.analog-stereo
[1600085690.454|15245|audiostream.cpp :156 ] Stream successfully created, connected to alsa_input.usb-GN_Netcom_A_S_Jabra_PRO_930_027608F15805-00.mono-fallback
[1600085690.454|15245|pulselayer.cpp :371 ] All streams ready, starting !
[1600085697.904|15229|manager.cpp :1665 ] [call:5190575654160253] Remove local audio
[1600085697.904|15229|ringbufferpool.cpp:242 ] Unbind call 5190575654160253 from all bound calls
[1600085697.904|15229|sipcall.cpp :368 ] [call:5190575654160253] Terminate SIP session
```
It seems that Jami has problem playing early media from SDP after 183 SIP message (opening sound device problem maybe?) and has not such problem when self playing ringback tone after 180 SIP message.
Sound configuration is pulseaudio, rindtone device = default (uses standard speakers), output and input devices = Jabra PRO 930 Mono and works ok in other functions (i.e. incomming ringing on standard speakers but ringback tones and speech on Jabra).
Regards,
Pawełhttps://git.jami.net/savoirfairelinux/jami-project/-/issues/847IPv6 for SIP unavailable2022-11-11T16:26:42ZPavel PolyakovIPv6 for SIP unavailableIt looks like Jami doesn't support IPv6, neither does Jami listen on IPv6 when it is started up or does it allow calling an IPv6 host (truncates the address then says Bad URI).
This is pretty sad knowing that SIP will usually perform be...It looks like Jami doesn't support IPv6, neither does Jami listen on IPv6 when it is started up or does it allow calling an IPv6 host (truncates the address then says Bad URI).
This is pretty sad knowing that SIP will usually perform better in IPv6 since hosts can have a public address of their own and establish direct connections without relying on clumsy NAT bypass techniques and port redirections.Sébastien BlinAntoine NoreauSébastien Blinhttps://git.jami.net/savoirfairelinux/jami-daemon/-/issues/185Call Forward* { on Busy ; No Answer ; Always } for Jami accounts2024-02-09T20:01:33ZovariCall Forward* { on Busy ; No Answer ; Always } for Jami accounts[Improving SIP Support for Ring](https://ring.cx/en/news#improving-sip-support-for-ring) shared news about the `Call Transfer` feature.
Please add the following:
* `Call Forward on Busy`: Forward all calls if you are on the phone to a n...[Improving SIP Support for Ring](https://ring.cx/en/news#improving-sip-support-for-ring) shared news about the `Call Transfer` feature.
Please add the following:
* `Call Forward on Busy`: Forward all calls if you are on the phone to a number of your choice. Standard call rates may apply.
* `Call Forward No Answer`: Forward all unanswered calls to a number of your choice. Standard call rates may apply.
* `Call Forward Always`: Forward all calls to your SIP (VoIP) phone number to another number of your choice. Standard call rates may apply.
![image](/uploads/4df8d0a3947eff09dd5747925ac97034/image.png)
The idea for this feature (which also shows icons in the image above) came from https://www.exetel.com.au/phone/voip-features
[**Difference Between Call Forward and Call Transfer**](https://learningnetwork.cisco.com/thread/85291)
<b><i>Call Forwarding</i></b>
Call forwarding allows you to send all your incoming calls to another landline or cell phone number. Call forwarding overrides the ability to answer the phone forwarded from the original line. This service requires a subscription through your phone service provider and may incur an additional monthly fee.
<b><i>Call Transfer</i></b>
Call transfer allows you to send a call from one phone to another telephone without the need to disconnect the phone call. This feature is usually activated by the push of a button followed by dialing an extension.
Thank youhttps://git.jami.net/savoirfairelinux/jami-daemon/-/issues/109SIP account TLS/SRTP configuration is old fashioned/confusing2022-11-11T16:13:35ZMaxim CournoyerSIP account TLS/SRTP configuration is old fashioned/confusingTested on: Android 20190103 (f-droid)
Recently, voip.ms (finally) added support for calls encryption using TLS for signaling and SRTP for media. I was thrilled to try it with Jami, but couldn't get it to work:
1. Add SIP account basic ...Tested on: Android 20190103 (f-droid)
Recently, voip.ms (finally) added support for calls encryption using TLS for signaling and SRTP for media. I was thrilled to try it with Jami, but couldn't get it to work:
1. Add SIP account basic detail (alias, hostname, username & password), registered OK.
2. Went to Security tab (android client), and enable TLS transport. As voip.ms is using a trusted SSL certificate, I wouldn't expect to have to do anything else, but:
a) the greyed out options below suggest that only the client certificate is going to be verified (I don't care about my cert, but I do want to authenticate the SIP server). So I checked the "Verify Server" box, and unchecked "Verify Client" and "TLS Require Client Certificate".
b) I have no idea why there's a "Server Name" field; this should at least defaults to my SIP hostname, if required?
c) There are other options which are nice for a self signed certs setup I guess, but overly complicated for the more straightforward CA signed use case. Perhaps they could be hidden under an "advanced" section?
d) I put my hostname in Server Name, just in case, and left the other options empty/default.
Expected result: SIP account is re-registered using TLS.
Actual result: TLS seems to fail silently, option is reverted to disabled when visiting the security menu.Sébastien BlinSébastien Blin