jami-project issueshttps://git.jami.net/savoirfairelinux/jami-project/-/issues2023-03-31T15:54:47Zhttps://git.jami.net/savoirfairelinux/jami-project/-/issues/1512First call: "Searching" remains if the Android contact is called after JUST t...2023-03-31T15:54:47ZElysFirst call: "Searching" remains if the Android contact is called after JUST the Android contact disables and enables (again) his own account (only useraccount1 -> useraccount2?)using Jami (Windows 10 -> Android; useraccount1 -> useraccount2), different network
EDIT: I think I didn't reproduce if I make a call **sameuseraccount -> sameuseraccount** if the Android user enables his account very quickly again - ho...using Jami (Windows 10 -> Android; useraccount1 -> useraccount2), different network
EDIT: I think I didn't reproduce if I make a call **sameuseraccount -> sameuseraccount** if the Android user enables his account very quickly again - however if the **Android user waits** one **minute or even longer** and THEN **enables** his account again I get the **"Connecting"** issue https://git.jami.net/savoirfairelinux/jami-project/-/issues/1513 (at least after I repeat these steps the second time: Open Jami (Windows 10 and Android); Wait a few seconds; Now just the Android user disables his own account and waits two minutes or even longer; Now the Android user enables his own account; Same steps: The Android user disables his own account and waits two minutes; Now the Android user enables his own account: This time a call sameuseraccount(windows 10) -> sameuseraccount(android) will stuck in "Connecting")
You have to make a call **useraccount1 -> useraccount2** if the Android user **immediately** enables his account
Mixed:
- If I first make a call sameuseraccount(windows 10) -> sameuseraccount(android) everything works
- Now I disable (Windows 10) sameuseraccount(windows 10) and enable the other account (useraccount1 (windows 10)) so I can make a call useraccount1 (Windows 10) -> useraccount2 (Android)
- Still same issue (although sameuseraccount(windows 10) -> sameuseraccount(android) does not have any issues?)
Maybe also related: https://git.jami.net/savoirfairelinux/jami-project/-/issues/1513
ALWAYS DISABLE "run in background"
Steps to reproduce:
1. Windows 10: (=useraccount1) You've already opened Jami and keep Jami running
1. Android (=useraccount2) Open Jami
1. Now the Windows 10 user will view the green dot of the Android user in the contact list
1. Now just the Android user disables his own account
1. Now just the Android user enables his own account
1. BEFORE the Android user views the green dot of (all) his contacts in the contact list the Windows 10 user tries to make a call (Windows 10 -> Android)
1. Windows 10: The call will be stuck at "Searching" (it won't fail? - it won't work?; even after a minute; even if the Android user now views the green dot of all contacts)
Sometimes after some time you'll get missed outgoing call (in that case Windows 10).
Same issue if you immediately make a **first call** (Windows 10 -> Android) if the Android user views the green dot of the Windows 10 user in the contact list
However - the **next calls will work**?
EDIT: I think this is not (only) a timing issue but an issue because **(just) the Android user disabled** (and now enables) his own account.
**No issues** if you make a call **Android -> Windows 10**. (Remember you never disabled the Windows 10 Jami account)
EDIT 2: If the Windows 10 user also disables his own account and enables his account again I think this issue does NOT happen? (Not sure - I also had this issue if the Windows 10 user disables and enables his account?)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,
Paweł