No Audio on specific device
Hi guys,
we are using pjsip2.5.5 on android devices since a few years now.
Now we have a problem with audio output on an AllNet Touch Display 10" running Android 6
Here is some output from starting pjsip and receiving a call with no audio out on device sip:166...
We are using Android JNI Audio device and opus codec, on all other devices we are using, this setup is working fine.
Do you have any ideas where the problem is? Is the call routed to some wrong output stream? Do we have to add anything to the conference bridge?
We have already tried to use opensl ES audio device, with no luck.
Thanks a lot
Flo
D/PJSUA Native: 20:42:36.348 sip_endpoint.c .Module "mod-pjsua-log" registered
D/PJSUA Native: 20:42:36.348 sip_endpoint.c .Module "mod-tsx-layer" registered
D/PJSUA Native: 20:42:36.348 sip_endpoint.c .Module "mod-stateful-util" registered
D/PJSUA Native: 20:42:36.348 sip_endpoint.c .Module "mod-ua" registered
D/PJSUA Native: 20:42:36.348 sip_endpoint.c .Module "mod-100rel" registered
D/PJSUA Native: 20:42:36.348 sip_endpoint.c .Module "mod-pjsua" registered
D/PJSUA Native: 20:42:36.348 sip_endpoint.c .Module "mod-invite" registered
D/PJSUA Native: 20:42:36.348 opensl_dev.c ..OpenSL sound library initialized
D/PJSUA Native: 20:42:36.348 android_jni_de ..Android JNI sound library initialized
D/PJSUA Native: 20:42:36.350 pjlib ..select() I/O Queue created (0xe07cbe14)
D/PJSUA Native: 20:42:36.358 sip_endpoint.c .Module "mod-evsub" registered
D/PJSUA Native: 20:42:36.358 sip_endpoint.c .Module "mod-presence" registered
D/PJSUA Native: 20:42:36.358 sip_endpoint.c .Module "mod-mwi" registered
D/PJSUA Native: 20:42:36.358 sip_endpoint.c .Module "mod-refer" registered
D/PJSUA Native: 20:42:36.358 sip_endpoint.c .Module "mod-pjsua-pres" registered
D/PJSUA Native: 20:42:36.358 sip_endpoint.c .Module "mod-pjsua-im" registered
D/PJSUA Native: 20:42:36.358 sip_endpoint.c .Module "mod-pjsua-options" registered
D/PJSUA Native: 20:42:36.359 pjsua_core.c .1 SIP worker threads created
I/PJSUA Native: 20:42:36.359 pjsua_core.c .pjsua version 2.5.5 for Linux-3.10.65/armv8l initialized
D/PJSUA Native: 20:42:36.359 pjsua_core.c .PJSUA state changed: CREATED --> INIT
I/PJSUA Native: Got 2 audio devices
I/PJSUA Native: 0. OpenSL ES Audio (in=1, out=1)
I/PJSUA Native: 1. Android JNI (in=1, out=1)
D/PJSUA Native: 20:42:36.359 pjsua_aud.c Set sound device: capture=1, playback=1
D/PJSUA Native: 20:42:36.359 pjsua_aud.c .Opening sound device (speaker + mic) PCM@16000/1/20ms
D/PJSUA Native: 20:42:36.359 android_jni_de ..Creating Android JNI stream
D/PJSUA Native: 20:42:36.364 android_jni_de ..Using audio input source : 7
D/PJSUA Native: 20:42:36.395 android_jni_de ..Audio record initialized successfully.
D/PJSUA Native: 20:42:36.396 android_jni_de Setting thread priority successful
D/PJSUA Native: 20:42:36.411 android_jni_de !..Audio track initialized successfully.
I/PJSUA Native: 20:42:36.411 echo_webrtc.c ..WebRTC AEC mobile successfully created with options 3
D/PJSUA Native: 20:42:36.412 ec0xe0838800 ..WebRTC AEC created, clock_rate=16000, channel=1, samples per frame=320, tail length=30 ms, latency=100 ms
D/PJSUA Native: 20:42:36.412 android_jni_de ..Android JNI stream started
I/PJSUA Native: setting sound device successful to 1 (0=opensl_es, 1=android_jni)
D/PJSUA Native: 20:42:36.413 android_jni_de Setting thread priority successful
D/PJSUA Native: 20:42:36.417 tlstp:46271 !SIP TLS listener is ready for incoming connections at 10.100.3.167:46271
D/PJSUA Native: 20:42:36.418 pjsua_core.c PJSUA state changed: INIT --> STARTING
D/PJSUA Native: 20:42:36.418 sip_endpoint.c .Module "mod-unsolicited-mwi" registered
D/PJSUA Native: 20:42:36.418 pjsua_core.c .PJSUA state changed: STARTING --> RUNNING
I/PJSUA Native: starting account configuration...
I/PJSUA Native: ... configs done, setting srtp to mandatory
->>> adding account Registration
...
<<<-
D/PJSUA Native: 20:42:37.419 pjsua_aud.c !Closing sound device after idle for 1 second(s)
D/PJSUA Native: 20:42:37.419 pjsua_aud.c .Closing Android JNI sound playback device and Android JNI sound capture device
D/PJSUA Native: 20:42:37.420 android_jni_de .Android JNI stream stopped
D/PJSUA Native: 20:42:37.420 android_jni_de .Destroying Android JNI stream...
D/PJSUA Native: 20:42:37.450 android_jni_de !.Audio record released
D/PJSUA Native: 20:42:37.461 android_jni_de .Audio track released
D/PJSUA Native: 20:42:37.461 android_jni_de .Android JNI stream destroyed
D/PJSUA Native: 20:43:04.914 pjsua_core.c .RX 1117 bytes Request msg INVITE/cseq=102 (rdata0xe07ef5e0) from TLS 10.100.3.81:5061:
INVITE sip:166@10.100.3.167:46910;transport=TLS;ob SIP/2.0
Via: SIP/2.0/TLS 10.100.3.81:5061;branch=z9hG4bK72589e35;rport
Max-Forwards: 70
From: "DU Flo" sip:93@10.100.3.81;tag=as17062081
To: sip:166@10.100.3.167:46910;transport=TLS;ob
Contact: sip:93@10.100.3.81:5061;transport=TLS
Call-ID: 5e4090b57056947a7c744e4b4c6f78f5@10.100.3.81:5061
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.7.0~dfsg-0~ppa1
Date: Thu, 03 May 2018 12:43:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-callom: 93,CallFromDU
Remote-Party-ID: "DU Flo" sip:93@10.100.3.81;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 379
v=0
o=root 1077773950 1077773950 IN IP4 10.100.3.81
s=Asterisk PBX 13.7.0~dfsg-0~ppa1
c=IN IP4 10.100.3.81
t=0 0
m=audio 19522 RTP/SAVP 107 101
a=rtpmap:107 opus/
D/PJSUA Native: 20:43:04.915 pjsua_call.c .Incoming Request msg INVITE/cseq=102 (rdata0xe07ef5e0)
D/PJSUA Native: 20:43:04.921 pjsua_media.c ..Call 0: initializing media..
D/PJSUA Native: 20:43:04.932 pjsua_media.c ...RTP socket reachable at 10.100.3.167:4000
D/PJSUA Native: 20:43:04.932 pjsua_media.c ...RTCP socket reachable at 10.100.3.167:4001
D/PJSUA Native: 20:43:04.932 pjsua_media.c ...Media index 0 selected for audio call 0
D/PJSUA Native: 20:43:04.934 pjsua_core.c .....TX 293 bytes Response msg 100/INVITE/cseq=102 (tdta0xe0ff5000) to TLS 10.100.3.81:5061:
SIP/2.0 100 Trying
Via: SIP/2.0/TLS 10.100.3.81:5061;rport=5061;received=10.100.3.81;branch=z9hG4bK72589e35
Call-ID: 5e4090b57056947a7c744e4b4c6f78f5@10.100.3.81:5061
From: "DU Flo" sip:93@10.100.3.81;tag=as17062081
To: sip:166@10.100.3.167;ob
CSeq: 102 INVITE
Content-Length: 0
--end msg--
I/PJSUA Native: Thread attached
I/PJSUA Native: 20:43:04.940 PJSUA Native ..Incoming call from "DU Flo" sip:93@10.100.3.81!!
I/PJSUA Native: 20:43:04.940 PJSUA Native ..Header X-callom: 93,CallFromDU
Remote-Party-ID: "DU Flo" sip:93@10.100.3.81;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 379
v=0
o=root 1077773950 1077773950 IN IP4 10.100.3.81
s=Asterisk PBX 13.7.0~dfsg-0~ppa1
c=IN IP4 10.100.3.81
t=0 0
m=audio 19522 RTP/SAVP 107 101
a=rtpmap:107 opus/48000/2
a=fmtp:107 useinbandfec=1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:60
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:QIsqyLJMAUnuINJip15j8J8pOlxCKHBPiq8JNKHo
D/PJSUA Native: 20:43:04.948 pjsua_call.c ..Answering call 0: code=180
D/PJSUA Native: 20:43:04.949 pjsua_core.c ......TX 489 bytes Response msg 180/INVITE/cseq=102 (tdta0xe0ff5000) to TLS 10.100.3.81:5061:
SIP/2.0 180 Ringing
Via: SIP/2.0/TLS 10.100.3.81:5061;rport=5061;received=10.100.3.81;branch=z9hG4bK72589e35
Call-ID: 5e4090b57056947a7c744e4b4c6f78f5@10.100.3.81:5061
From: "DU Flo" sip:93@10.100.3.81;tag=as17062081
To: sip:166@10.100.3.167;ob;tag=32415dfc-263d-4162-8863-1703a422eaec
CSeq: 102 INVITE
Contact: sip:166@10.100.3.167:46910;transport=TLS;ob
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length: 0
--end msg--
I/PJSUA Native: 20:43:04.949 CallState .........CallState = 180
I/PJSUA Native: 20:43:04.949 CallState .........Call 0 state=EARLY
I/PJSUA Native: 20:43:04.949 CallState .........CallID = 5e4090b57056947a7c744e4b4c6f78f5@10.100.3.81:5061
I/PJSUA Native: 20:43:04.949 CallState .........local = sip:166@10.100.3.167;ob
I/PJSUA Native: 20:43:04.949 CallState .........remote = "DU Flo" sip:93@10.100.3.81
D/PJSUA Native: 20:43:08.534 pjsua_call.c !Answering call 0: code=200
D/PJSUA Native: 20:43:08.534 pjsua_media.c ...Call 0: updating media..
D/PJSUA Native: 20:43:08.535 pjsua_aud.c ....Audio channel update..
D/PJSUA Native: 20:43:08.535 strm0xde006814 .....Encoder stream started
D/PJSUA Native: 20:43:08.535 strm0xde006814 .....Decoder stream started
D/PJSUA Native: 20:43:08.536 pjsua_media.c ....Audio updated, stream #0: opus (sendrecv)
I/PJSUA Native: PJSUA_CALL_MEDIA_STATE_CHANGE
I/PJSUA Native: PJSUA_CALL_MEDIA_ACTIVE
I/PJSUA Native: 20:43:08.536 MEDIA_ACTIVE ...ci.conf_slot = 1
I/PJSUA Native: 20:43:08.536 PortInfo ...slot_id=1
I/PJSUA Native: 20:43:08.536 PortInfo ...name=sip:93@10.100.3.81
I/PJSUA Native: 20:43:08.536 PortInfo ...clock_rate=48000
I/PJSUA Native: 20:43:08.536 PortInfo ...channel_count=1
I/PJSUA Native: 20:43:08.536 PortInfo ...samples_per_frame=960
I/PJSUA Native: 20:43:08.536 PortInfo ...bits_per_sample=16
I/PJSUA Native: 20:43:08.536 PortInfo ...number_of_listeners=0
I/PJSUA Native: 20:43:08.536 PortInfo ...remote info="DU Flo" sip:93@10.100.3.81
I/PJSUA Native: 20:43:08.536 PortInfo ...remote contact=sip:93@10.100.3.81:5061;transport=TLS
I/PJSUA Native: 20:43:08.536 MediaInfo ...clock_rate=16000
I/PJSUA Native: 20:43:08.540 MediaInfo ...snd_clock_rate=0
I/PJSUA Native: 20:43:08.540 MediaInfo ...ec_options=3
I/PJSUA Native: 20:43:08.541 MediaInfo ...no_vad=1
I/PJSUA Native: 20:43:08.541 MediaInfo ...snd_play_latency=140
I/PJSUA Native: 20:43:08.541 MediaInfo ...snd_rec_latency=100
I/PJSUA Native: 20:43:08.541 MediaInfo ...ec_tail_len=30
I/PJSUA Native: 20:43:08.541 MediaInfo ...quality=10
D/PJSUA Native: 20:43:08.541 pjsua_aud.c ...Conf connect: 1 --> 0
D/PJSUA Native: 20:43:08.541 pjsua_aud.c ....Set sound device: capture=1, playback=1
D/PJSUA Native: 20:43:08.541 pjsua_aud.c .....Opening sound device (speaker + mic) PCM@16000/1/20ms
D/PJSUA Native: 20:43:08.541 android_jni_de ......Creating Android JNI stream
D/PJSUA Native: 20:43:08.543 android_jni_de ......Using audio input source : 7
D/PJSUA Native: 20:43:08.554 android_jni_de ......Audio record initialized successfully.
D/PJSUA Native: 20:43:08.561 android_jni_de ......Audio track initialized successfully.
I/PJSUA Native: 20:43:08.561 echo_webrtc.c ......WebRTC AEC mobile successfully created with options 3
D/PJSUA Native: 20:43:08.561 ec0xde4f6100 ......WebRTC AEC created, clock_rate=16000, channel=1, samples per frame=320, tail length=30 ms, latency=100 ms
D/PJSUA Native: 20:43:08.561 android_jni_de ......Android JNI stream started
D/PJSUA Native: 20:43:08.561 conference.c ....Port 1 (sip:93@10.100.3.81) transmitting to port 0 (Android JNI)
D/PJSUA Native: 20:43:08.561 pjsua_aud.c ...Conf connect: 0 --> 1
D/PJSUA Native: 20:43:08.562 conference.c ....Port 0 (Android JNI) transmitting to port 1 (sip:93@10.100.3.81)
D/PJSUA Native: 20:43:08.562 pjsua_core.c ....TX 960 bytes Response msg 200/INVITE/cseq=102 (tdta0xe0ff5000) to TLS 10.100.3.81:5061:
SIP/2.0 200 OK
Via: SIP/2.0/TLS 10.100.3.81:5061;rport=5061;received=10.100.3.81;branch=z9hG4bK72589e35
Call-ID: 5e4090b57056947a7c744e4b4c6f78f5@10.100.3.81:5061
From: "DU Flo" sip:93@10.100.3.81;tag=as17062081
To: sip:166@10.100.3.167;ob;tag=32415dfc-263d-4162-8863-1703a422eaec
CSeq: 102 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Contact: sip:166@10.100.3.167:46910;transport=TLS;ob
Supported: replaces, 100rel, timer, norefersub
Content-Type: application/sdp
Content-Length: 394
v=0
o=- 3734340184 3734340185 IN IP4 10.100.3.167
s=pjmedia
b=AS:117
t=0 0
a=X-nat:0
m=audio 4000 RTP/SAVP 107 101
c=IN IP4 10.100.3.167
b=TIAS:96000
a=rtcp:4001 IN IP4 10.100.3.167
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:Ku0atFGguQS2vamRCD2cnt2QsXivCsLHF+QaSAwr
a=rtpmap:107 opus/48000/2
a=fmtp:107 useinbandf
I/PJSUA Native: 20:43:08.562 CallState .......CallState = 200
I/PJSUA Native: 20:43:08.562 CallState .......Call 0 state=CONNECTING
I/PJSUA Native: 20:43:08.563 CallState .......CallID = 5e4090b57056947a7c744e4b4c6f78f5@10.100.3.81:5061
I/PJSUA Native: 20:43:08.563 CallState .......local = sip:166@10.100.3.167;ob
I/PJSUA Native: 20:43:08.563 CallState .......remote = "DU Flo" sip:93@10.100.3.81
D/PJSUA Native: 20:43:08.566 android_jni_de Setting thread priority successful
D/PJSUA Native: 20:43:08.568 pjsua_core.c .RX 474 bytes Request msg ACK/cseq=102 (rdata0xe07ef5e0) from TLS 10.100.3.81:5061:
ACK sip:166@10.100.3.167:46910;transport=TLS;ob SIP/2.0
Via: SIP/2.0/TLS 10.100.3.81:5061;branch=z9hG4bK1723ea35;rport
Max-Forwards: 70
From: "DU Flo" sip:93@10.100.3.81;tag=as17062081
To: sip:166@10.100.3.167:46910;transport=TLS;ob;tag=32415dfc-263d-4162-8863-1703a422eaec
Contact: sip:93@10.100.3.81:5061;transport=TLS
Call-ID: 5e4090b57056947a7c744e4b4c6f78f5@10.100.3.81:5061
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.7.0~dfsg-0~ppa1
Content-Length: 0
--end msg--
D/PJSUA Native: 20:43:08.575 Master/sound !Underflow, buf_cnt=0, will generate 1 frame
I/PJSUA Native: 20:43:08.580 CallState !...CallState = 200
I/PJSUA Native: 20:43:08.580 CallState ...Call 0 state=CONFIRMED
I/PJSUA Native: 20:43:08.580 CallState ...CallID = 5e4090b57056947a7c744e4b4c6f78f5@10.100.3.81:5061
I/PJSUA Native: 20:43:08.580 CallState ...local = sip:166@10.100.3.167;ob
I/PJSUA Native: 20:43:08.580 CallState ...remote = "DU Flo" sip:93@10.100.3.81
D/PJSUA Native: 20:43:08.580 Master/sound !Underflow, buf_cnt=0, will generate 1 frame
D/PJSUA Native: 20:43:08.585 Master/sound !Underflow, buf_cnt=0, will generate 1 frame
D/PJSUA Native: 20:43:08.587 android_jni_de !Setting thread priority successful
D/PJSUA Native: 20:43:08.592 Master/sound !Underflow, buf_cnt=0, will generate 1 frame
D/PJSUA Native: 20:43:08.717 Master/sound !Underflow, buf_cnt=0, will generate 1 frame
D/PJSUA Native: 20:43:08.722 Master/sound Underflow, buf_cnt=0, will generate 1 frame
D/PJSUA Native: 20:43:08.741 Master/sound Underflow, buf_cnt=0, will generate 1 frame
D/PJSUA Native: 20:43:08.752 Master/sound Underflow, buf_cnt=0, will generate 1 frame
D/PJSUA Native: 20:43:08.904 Master/sound Underflow, buf_cnt=0, will generate 1 frame
D/PJSUA Native: 20:43:22.750 pjsua_aud.c !Conf disconnect: 1 -x- 0
D/PJSUA Native: 20:43:22.750 conference.c .Port 1 (sip:93@10.100.3.81) stop transmitting to port 0 (Android JNI)
D/PJSUA Native: 20:43:22.750 pjsua_aud.c Conf disconnect: 0 -x- 1
D/PJSUA Native: 20:43:22.750 conference.c .Port 0 (Android JNI) stop transmitting to port 1 (sip:93@10.100.3.81)
D/PJSUA Native: 20:43:22.750 pjsua_call.c Call 0 hanging up: code=0..
D/PJSUA Native: 20:43:22.751 pjsua_core.c ....TX 394 bytes Request msg BYE/cseq=31380 (tdta0xe184a000) to TLS 10.100.3.81:5061:
BYE sip:93@10.100.3.81:5061;transport=TLS SIP/2.0
Via: SIP/2.0/TLS 10.100.3.167:46910;rport;branch=z9hG4bKPjc2e3d2eb-2aaa-442c-9811-d17d9e7b9585;alias
Max-Forwards: 70
From: sip:166@10.100.3.167;ob;tag=32415dfc-263d-4162-8863-1703a422eaec
To: "DU Flo" sip:93@10.100.3.81;tag=as17062081
Call-ID: 5e4090b57056947a7c744e4b4c6f78f5@10.100.3.81:5061
CSeq: 31380 BYE
Content-Length: 0
--end msg--