Hi all, It's been more than a week that I'm trying to get PJSIP under Android NDK working. As first step,I followed Android Getting-Started Guide and I based on the sample app provided under r5609: $PJSIP_DIR/pjsip-apps/src/swig/java/android/ I checked the creation of the libpjsip2.so and that is available under jniLibs/armeabi in the Android project. Other settings that I used were: Configuration: NDK_TOOLCHAIN_VERSION=4.9 APP_PLATFORM=android-21 TARGET_ABI=armeabi ./configure-android --use-ndk-cflags Min SDK: 16, Target-SDK: 21 (Manifest and Gradle files) I run the app and give the SIP Configurations (Asterisk SIP server) as follows: ID: sip:<client-nr>@<sip-server-ip> Registrar: sip:<sip-server-ip> Proxy: Username: <client-nr> Password: **** The registration succeeds as: pjsua_core.c .RX 613 bytes Response msg 200/REGISTER/cseq=46791 (rdata0xe593d014) from UDP org.pjsip.pjsua2.app I/System.out: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.16.90.224:6000 ;branch=z9hG4bKPjb8ad6948-ec21-43c1-b386-851248b5faaf;received=10.16.90.224;rport=6000 From: <sip:8090404@10.16.90.13>;tag=a6c0cdd0-a1cd-431e-a81b-7155506858ca To: <sip:8090404@10.16.90.13>;tag=as6c12fc38 Call-ID: e9646757-dbb2-4d62-ae2c-79dd7d417602 CSeq: 46791 REGISTER Server: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 300 Contact: <sip:8090404@10.16.90.224:6000;ob>;expires=300 Date: Wed, 28 Jun 2017 14:09:20 GMT Content-Length: 0 --end msg-- 13:54:06.640 pjsua_acc.c ....SIP outbound status for acc 0 is not active 13:54:06.642 pjsua_acc.c ....sip:8090404@10.16.90.13: registration success, status=200 (OK), will re-register in 300 seconds 13:54:06.644 pjsua_acc.c ....Keep-alive timer started for acc 0, destination:10.16.90.13:5060, interval:15s org.pjsip.pjsua2.app I/ViewRootImpl: ViewRoot's Touch Event : ACTION_DOWN Hoever when I receive calls from other parties registered in the SIP system, the call is not shown until the other party hangs up the phone. The logs looks as following: 13:54:37.828 pjsua_core.c .RX 855 bytes Request msg INVITE/cseq=102 (rdata0xe593d014) from UDP 10.16.90.13:5060: INVITE sip:8090404@10.16.90.224:6000;ob SIP/2.0 Via: SIP/2.0/UDP 10.16.90.13:5060;branch=z9hG4bK7bb6182a;rport Max-Forwards: 70 From: "ZuB mobile 8" <sip:8090402@10.16.90.13>;tag=as36e6e307 To: <sip:8090404@10.16.90.224:6000;ob> Contact: <sip:8090402@10.16.90.13:5060> Call-ID: 37bdcb3d69838c761f0397767c1648fa@10.16.90.13:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u4 Date: Wed, 28 Jun 2017 14:09:51 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 250 v=0 o=root 1541597689 1541597689 IN IP4 10.16.90.13 s=Asterisk PBX 1.8.13.1~dfsg1-3+deb7u4 c=IN IP4 10.16.90.13 t=0 0 m=audio 12074 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --end msg-- 13:54:37.837 pjsua_call.c .Incoming Request msg INVITE/cseq=102 (rdata0xe593d014) 13:54:37.840 pjsua_media.c ..Call 0: initializing media.. 13:54:37.840 pjsua_core.c ...Trying STUN server stun.pjsip.org (1 of 1).. 13:54:45.745 utsx0xd5c5ef50 ...STUN timeout waiting for response 13:54:45.748 stunresolve ....Session failed because STUN Binding request failed: STUN transaction has timed out (PJNATH_ESTUNTIMEDOUT) 13:54:45.749 pjsua_core.c ....STUN resolution for stun.pjsip.org failed: STUN transaction has timed out (PJNATH_ESTUNTIMEDOUT) 13:54:45.751 pjsua_core.c ....STUN resolution failed: STUN transaction has timed out (PJNATH_ESTUNTIMEDOUT) 13:54:45.805 pjsua_core.c ...Ignoring STUN resolution failure (by setting) 13:54:45.807 icetp00 ...Creating ICE stream transport with 2 component(s) 13:54:45.812 icetp00 ....Comp 1/0: host candidate 10.16.90.224:37054 (tpid=64) added 13:54:45.814 icetp00 ....Comp 2/0: host candidate 10.16.90.224:43063 (tpid=64) added 13:54:45.815 icetp00 ....ICE stream transport 0xd5c74614 created 13:54:45.815 pjsua_media.c ...Media index 0 selected for audio call 0 13:54:45.816 pjsua_core.c .....TX 308 bytes Response msg 100/INVITE/cseq=102 (tdta0xd5c1e064) to UDP 10.16.90.13:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.16.90.13:5060 ;rport=5060;received=10.16.90.13;branch=z9hG4bK7bb6182a Call-ID: 37bdcb3d69838c761f0397767c1648fa@10.16.90.13:5060 From: "ZuB mobile 8" <sip:8090402@10.16.90.13>;tag=as36e6e307 To: <sip:8090404@10.16.90.224;ob> CSeq: 102 INVITE Content-Length: 0 --end msg-- ======== Incoming call ======== 13:54:45.818 pjsua_call.c !Answering call 0: code=180 13:54:45.818 pjsua_call.c .Pending answering call 0 upon completion of media transport Activity_launch_request id:org.pjsip.pjsua2.app time:8914524 13:54:45.817 pjsua_core.c .RX 855 bytes Request msg INVITE/cseq=102 (rdata0xe593d014) from UDP 10.16.90.13:5060: INVITE sip:8090404@10.16.90.224:6000;ob SIP/2.0 Via: SIP/2.0/UDP 10.16.90.13:5060;branch=z9hG4bK7bb6182a;rport Max-Forwards: 70 From: "ZuB mobile 8" <sip:8090402@10.16.90.13>;tag=as36e6e307 To: <sip:8090404@10.16.90.224:6000;ob> Contact: <sip:8090402@10.16.90.13:5060> Call-ID: 37bdcb3d69838c761f0397767c1648fa@10.16.90.13:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u4 Date: Wed, 28 Jun 2017 14:09:51 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 250 v=0 o=root 1541597689 1541597689 IN IP4 10.16.90.13 s=Asterisk PBX 1.8.13.1~dfsg1-3+deb7u4 c=IN IP4 10.16.90.13 t=0 0 m=audio 12074 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --end msg-- At last I cannot place calls. They never come/appear on the handset of the other party. On the other parties I have a funcitoning SIP - Client App, hence they are in place to make and receive SIP calls. In the following the Android logcat I can see that the call is initiated but cannot be really placed later. 14:28:16.793 pjsua_call.c Making call with acc #0 to sip:8090404@10.16.90.13 14:28:16.793 pjsua_aud.c .Set sound device: capture=-1, playback=-2 14:28:16.793 pjsua_aud.c ..Opening sound device (speaker + mic) PCM@16000/1/20ms 14:28:16.793 android_jni_de ...Creating Android JNI stream getMinFrameCount 640 getIoDescriptor: ioHandle = 21, index = 1, mIoDescriptors = 0xd5c65c38 getSamplingRate() ioHandle 21, sampling rate 48000 getIoDescriptor: ioHandle = 21, index = 1, mIoDescriptors = 0xd5c65c38 getFrameCount() ioHandle 21, frameCount 1920 getIoDescriptor: ioHandle = 21, index = 1, mIoDescriptors = 0xd5c65c38 getLatency() output 21, latency 80 calculateMinFrameCount afLatency 80 afFrameCount 1920 afSampleRate 48000 sampleRate 16000 speed 1.000000 minBufCount: 2 V/AudioTrack: getMinFrameCount=1288: afFrameCount=1920, afSampleRate=48000, afLatency=80 14:28:16.798 android_jni_de ...Using audio input source : 7 set(): inputSource 7, sampleRate 16000, format 0x1, channelMask 0x10, frameCount 640, notificationFrames 0, sessionId 0, transferType 0, flags 0, opPackageName org.pjsip.pjsua2.app uid -1, pid -1 Building AudioRecord with attributes: source=7 flags=0x0 tags=[] set(): mSessionId 969 set: Create AudioRecordThread getIoDescriptor: ioHandle = 286, index = -2, mIoDescriptors = 0xd5c65c38 ioConfigChanged: [Update mIoDescriptors] add ioHandle = 286 -> descriptor = 0xe5d17734 ioConfigChanged() new input opened 286 samplingRate 16000, format 0x1 channel mask 0x10 frameCount 320 deviceId 0 getIoDescriptor: ioHandle = 286, index = 3, mIoDescriptors = 0xd5c65c38 getSamplingRate() ioHandle 286, sampling rate 16000 14:28:16.814 android_jni_de ...Audio record initialized successfully. 14:28:16.815 android_jni_de !Setting thread priority successful start, sync event 0 trigger session 0 mAudioRecord->start() set(): streamType -1, sampleRate 16000, format 0x1, channelMask 0x1, frameCount 1288, flags #0, notificationFrames 0, sessionId 0, transferType 3, uid -1, pid -1 set() streamType -1 frameCount 1288 flags 0000 Building AudioTrack with attributes: usage=2 content=1 flags=0x0 tags=[] set: Create AudioTrackThread getIoDescriptor: ioHandle = 286, index = 3, mIoDescriptors = 0xd5c65c38 ioConfigChanged() new config for input 286 samplingRate 16000, format 0x1 channel mask 0x10 frameCount 320 frameCountHAL 320 deviceId 7 etIoDescriptor: ioHandle = 13, index = 0, mIoDescriptors = 0xd5c65c38 getLatency() output 13, latency 48 createTrack_l() output 13 afLatency 48 getIoDescriptor: ioHandle = 13, index = 0, mIoDescriptors = 0xd5c65c38 getFrameCount() ioHandle 13, frameCount 960 getIoDescriptor: ioHandle = 13, index = 0, mIoDescriptors = 0xd5c65c38 getFrameCountHAL() ioHandle 13, frameCount 192 getIoDescriptor: ioHandle = 13, index = 0, mIoDescriptors = 0xd5c65c38 getSamplingRate() ioHandle 13, sampling rate 48000 calculateMinFrameCount afLatency 48 afFrameCount 960 afSampleRate 48000 sampleRate 16000 speed 1.000000 minBufCount: 2 this(0xc6437aa0), mCblk(0xc7c80000), front(0), mIsOut 0, interrupt() FUTEX_WAKE D/AudioRecord: AudioRecord->stop() Client defaulted notificationFrames to 429 for frameCount 1288 14:28:16.838 android_jni_de !...Audio track initialized successfully. 14:28:16.840 ec0xc649d500 ...AEC created, clock_rate=16000, channel=1, samples per frame=320, tail length=200 ms, latency=0 ms 14:28:16.840 android_jni_de ...Android JNI stream started 14:28:16.842 pjsua_media.c .Call 2: initializing media.. 14:28:16.842 pjsua_core.c ..Trying STUN server stun.pjsip.org (1 of 1).. start, sync event 0 trigger session 0 mAudioRecord->start() 14:28:16.845 android_jni_de !Setting thread priority successful start(): 0xc6403600 getIoDescriptor: ioHandle = 13, index = 0, mIoDescriptors = 0xd5c65c38 ioConfigChanged() new config for output 13 samplingRate 48000, format 0x5 channel mask 0x3 frameCount 960 frameCountHAL 192 deviceId 1 14:28:19.741 pjsua_core.c !.RX 572 bytes Request msg OPTIONS/cseq=102 (rdata0xe593d014) from UDP 10.16.90.13:5060: OPTIONS sip:8090404@10.16.90.224:6000;ob SIP/2.0 Via: SIP/2.0/UDP 10.16.90.13:5060;branch=z9hG4bK61100e51;rport Max-Forwards: 70 From: "asterisk" <sip:asterisk@10.16.90.13>;tag=as41eca3c3 To: <sip:8090404@10.16.90.224:6000;ob> Contact: <sip:asterisk@10.16.90.13:5060> Call-ID: 3424500427a340620363ba0421441046@10.16.90.13:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u4 Date: Wed, 28 Jun 2017 14:43:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 .... 14:28:29.706 tcpc0xc6451814 TCP connect() error: Connection refused [code=120111] 14:28:29.707 tsx0xe594a064 Temporary failure in sending Request msg INVITE/cseq=2291 (tdta0xd5c1e064), will try next server: Connection refused 14:28:29.707 pjsua_core.c TX 1458 bytes Request msg INVITE/cseq=2291 (tdta0xd5c1e064) to UDP 10.16.90.13:5060: INVITE sip:8090404@10.16.90.13 SIP/2.0 Via: SIP/2.0/UDP 10.16.90.224:6000 ;rport;branch=z9hG4bKPjcec14480-39f2-4b56-9c6a-24d709db0644 Max-Forwards: 70 From: sip:8090404@10.16.90.13;tag=25df7888-cca1-4590-9a3d-0c564305c5a9 To: sip:8090404@10.16.90.13 Contact: <sip:8090404@10.16.90.224:6000;ob> Call-ID: df757a02-4bf9-4236-b9be-daae0d8a7ee7 CSeq: 2291 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: Pjsua2 Android 2.6-svn Authorization: Digest username="8090404", realm="asterisk", nonce="10191c35", uri="sip:8090404@10.16.90.13", response="e1596f80d122fa5162b3898680143441", algorithm=MD5 Content-Type: application/sdp Content-Length: 651 v=0 o=- 3707641709 3707641709 IN IP4 10.16.90.224 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 41449 RTP/AVP 98 97 99 104 3 0 8 9 96 c=IN IP4 10.16.90.224 b=TIAS:64000 a=rtcp:35730 IN IP4 10.16.90.224 a=sendrecv a=rtpmap:98 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:99 speex/32000 a=rtpmap:104 iLBC/8000 a=fmtp:104 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=ice-ufrag:43d7fa21 a=ice-pwd:506ecc17 a=candidate:Ha105ae0 1 UDP 2130706431 10.16.90.224 41449 typ host a=candidate:Ha105ae0 2 UDP 2130706430 10.16.90.224 35730 typ host When I build and check the sample app using my mac-book (I simply followed MacOS/Linux/BSD Getting-Started Guide) I could run the sample app while I could replace and answer SIP callls. Can you please give me any hints if you have any. Or do you see any problem at the current android sample. Thanks a lot in advance, Cheers Tunç