var sipster = require('sipster');
var srv = '192.168.0.234';
var ext = '1010';
var pwd = 'passw0rd';
// initialize pjsip
sipster.init();
// set up a transport to listen for incoming connections, defaults to UDP
var transport = new sipster.Transport({ port: 5060 });
var acc = new sipster.Account({
idUri: 'sip:' + ext + '@' + srv
, regConfig: {
registrarUri: 'sip:' + srv
}
, sipConfig: {
authCreds:[{
scheme:'digest'
, realm: '*'
, username: ext
, dataType: 0
, data: pwd
}]
}
});
acc.setRegistration(true);
acc.on('state', function(active, statusCode) {
console.log ("STATUS:" + statusCode);
});
acc.on('call', function(info, call) {
console.log('=== Incoming call from ' + info.remoteContact);
// watch for call state changes
call.on('state', function(state) {
console.log('=== Call state is now: ' + state.toUpperCase());
});
// listen for DTMF digits
call.on('dtmf', function(digit) {
console.log('=== DTMF digit received: ' + digit);
});
/*****
// audio stream(s) available
call.on('media', function(medias) {
// play looping .wav file to the first audio stream
var player = sipster.createPlayer('sound.wav');
player.startTransmitTo(medias[0]);
// record the audio of the other side, this will not include the audio from
// the player above.
var recorder = sipster.createRecorder('call.wav');
medias[0].startTransmitTo(recorder);
// to include the player audio, you can mix the sources together simply
// by transmitting to the same recorder:
// player.startTransmitTo(recorder);
});
***/
// answer the call (with default 200 OK)
call.answer();
});
sipster.start();
// acc.makeCall('sip:1000@192.168.0.234');
在接听或拨打电话时听不到声音，这是一个日志：
这是一些输出。我已经删除了大多数SIP对话，因为我认为与该问题无关。 我注意到音频通道似乎正在工作，PJSIP找到了ALSA驱动程序和相关设备
root@raspberrypi:~# node t1.js
18:55:41.206 os_core_unix.c !pjlib 2.8-svn for POSIX initialized
18:55:41.208 sip_endpoint.c .Creating endpoint instance...
18:55:41.208 pjlib .select() I/O Queue created (0x3b33770)
18:55:41.208 sip_endpoint.c .Module "mod-msg-print" registered
18:55:41.208 sip_transport. .Transport manager created.
18:55:41.208 pjsua_core.c .PJSUA state changed: NULL --> CREATED
18:55:41.208 sip_endpoint.c .Module "mod-pjsua-log" registered
18:55:41.209 sip_endpoint.c .Module "mod-tsx-layer" registered
18:55:41.209 sip_endpoint.c .Module "mod-stateful-util" registered
18:55:41.209 sip_endpoint.c .Module "mod-ua" registered
18:55:41.209 sip_endpoint.c .Module "mod-100rel" registered
18:55:41.209 sip_endpoint.c .Module "mod-pjsua" registered
18:55:41.209 sip_endpoint.c .Module "mod-invite" registered
18:55:41.302 alsa_dev.c ..ALSA driver found 14 devices
18:55:41.302 alsa_dev.c ..ALSA initialized
18:55:41.302 pjlib ..select() I/O Queue created (0x3b4f64c)
18:55:41.314 sip_endpoint.c .Module "mod-evsub" registered
18:55:41.314 sip_endpoint.c .Module "mod-presence" registered
18:55:41.314 sip_endpoint.c .Module "mod-mwi" registered
18:55:41.314 sip_endpoint.c .Module "mod-refer" registered
18:55:41.314 sip_endpoint.c .Module "mod-pjsua-pres" registered
18:55:41.314 sip_endpoint.c .Module "mod-pjsua-im" registered
18:55:41.314 sip_endpoint.c .Module "mod-pjsua-options" registered
18:55:41.314 pjsua_core.c .1 SIP worker threads created
18:55:41.314 pjsua_core.c .pjsua version 2.8-svn for Linux-4.19.42/armv7l/glibc-2.24 initialized
18:55:41.314 pjsua_core.c .PJSUA state changed: CREATED --> INIT
18:55:41.314 pjsua_aud.c Setting null sound device..
18:55:41.315 pjsua_aud.c .Opening null sound device..
18:55:41.316 pjsua_core.c SIP UDP socket reachable at 192.168.0.24:5060
18:55:41.316 udp0x3b72aa0 SIP UDP transport started, published address is 192.168.0.24:5060
18:55:41.317 pjsua_acc.c Adding account: id=sip:1010@192.168.0.234
18:55:41.317 pjsua_acc.c .Account sip:1010@192.168.0.234 added with id 0
18:55:41.317 pjsua_acc.c Acc 0: setting registration..
18:55:41.318 pjsua_core.c ..TX 485 bytes Request msg
REGISTER/cseq=37939 (tdta0x3b76e94) to UDP 192.168.0.234:5060:
REGISTER sip:192.168.0.234 SIP/2.0 ...
--end msg--
18:55:41.318 pjsua_acc.c .Acc 0: Registration sent
18:55:41.318 pjsua_core.c PJSUA state changed: INIT --> STARTING
18:55:41.319 sip_endpoint.c .Module "mod-unsolicited-mwi" registered
18:55:41.319 pjsua_core.c .PJSUA state changed: STARTING --> RUNNING
18:55:41.325 pjsua_core.c .RX 648 bytes Response msg 401/REGISTER/cseq=37939 (rdata0x3b740d4) from UDP 192.168.0.234:5060:
...
--end msg--
18:55:41.330 pjsua_core.c ....TX 744 bytes Request msg REGISTER/cseq=37940 (tdta0x3b76e94) to UDP 192.168.0.234:5060:
REGISTER sip:192.168.0.234 SIP/2.0
...
--end msg--
18:55:41.338 pjsua_core.c .RX 608 bytes Response msg 200/REGISTER/cseq=37940 (rdata0x7160129c) from UDP 192.168.0.234:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.24:5060;rport=5060;branch=z9hG4bKPj.XicpgD7rZ5URstdu899IaGdWZhFUOiC
--end msg--
18:55:41.343 pjsua_acc.c ....SIP outbound status for acc 0 is not active
18:55:41.343 pjsua_acc.c ....sip:1010@192.168.0.234: registration success, status=200 (OK), will re-register in 300 seconds
18:55:41.343 pjsua_acc.c ....Keep-alive timer started for acc 0, destination:192.168.0.234:5060:15, interval:1s
18:55:41.370 pjsua_core.c .RX 908 bytes Request msg NOTIFY/cseq=5065491 (rdata0x7160129c) from UDP 192.168.0.234:5060:
NOTIFY sip:1010@192.168.0.24:5060;ob SIP/2.0
...
--end msg--
18:55:41.376 pjsua_pres.c .Got unsolicited NOTIFY from 192.168.0.234:5060..
18:55:41.377 pjsua_core.c ...TX 306 bytes Response msg 200/NOTIFY/cseq=5065491 (tdta0x71603554) to UDP 192.168.0.234:5060:
SIP/2.0 200 OK
....
--end msg--
Registering ...
... Registered
STATUS:200
18:55:42.318 pjsua_aud.c Closing sound device after idle for 1 second(s)
18:55:42.318 pjsua_aud.c .Closing null sound device..
18:55:47.522 pjsua_core.c .RX 1281 bytes Request msg
从普通分机开始打入电话：
INVITE/cseq=5065494 (rdata0x7160129c) from UDP 192.168.0.234:5060:
INVITE sip:1010@192.168.0.24:5060;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.0.234;rport;branch=z9hG4bKDBpje8S4amrar
....
v=0
o=FreeSWITCH 1559210876 1559210877 IN IP4 192.168.0.234
s=FreeSWITCH
c=IN IP4 192.168.0.234
t=0 0
m=audio 27696 RTP/AVP 9 0 8 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
--end msg--
18:55:47.522 pjsua_call.c .Incoming Request msg INVITE/cseq=5065494 (rdata0x7160129c)
18:55:47.524 pjsua_media.c ..Call 0: initializing media..
18:55:47.525 pjsua_media.c ...RTP socket reachable at 192.168.0.24:4000
18:55:47.525 pjsua_media.c ...RTCP socket reachable at 192.168.0.24:4001
18:55:47.525 pjsua_media.c ...Media index 0 selected for audio call 0
=== Incoming call from <sip:mod_sofia@192.168.0.234:5060>
18:55:47.527 pjsua_call.c !Answering call 0: code=200
18:55:47.528 pjsua_core.c .....TX 302 bytes Response msg 100/INVITE/cseq=5065494 (tdta0x716128f4) to UDP 192.168.0.234:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.234;rport=5060;received=192.168.0.234;branch=z9hG4bKDBpje8S4amrar
Call-ID: 1f1855d2-fda6-1237-62be-00fd45fc4abc
From: "Interno 1000" <sip:1000@192.168.0.234>;tag=3yBB1QFK9vBDr
To: <sip:1010@192.168.0.24;ob>
CSeq: 5065494 INVITE
Content-Length: 0
--end msg--
18:55:47.531 inv0x716008dc ..SDP negotiation done: Success
18:55:47.531 pjsua_media.c ...Call 0: updating media..
18:55:47.531 pjsua_media.c .....Media stream call00:0 is destroyed
18:55:47.531 pjsua_aud.c ....Audio channel update..
18:55:47.531 strm0x3b76e44 .....VAD temporarily disabled
18:55:47.532 strm0x3b76e44 .....Encoder stream started
18:55:47.532 strm0x3b76e44 .....Decoder stream started
18:55:47.532 pjsua_media.c ....Audio updated, stream #0: G722 (sendrecv)
18:55:47.532 pjsua_core.c ....TX 873 bytes Response msg
根据代码，通话将由sipster应答，但是没有声音。
200/INVITE/cseq=5065494 (tdta0x3b718f4) to UDP 192.168.0.234:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.234;rport=5060;received=192.168.0.234;branch=z9hG4bKDBpje8S4amrar
Call-ID: 1f1855d2-fda6-1237-62be-00fd45fc4abc
From: "Interno 1000" <sip:1000@192.168.0.234>;tag=3yBB1QFK9vBDr
To: <sip:1010@192.168.0.24;ob>;tag=GlRjK-.n6i7GffuemwGd5ELoZq3icvj4
CSeq: 5065494 INVITE
Contact: <sip:1010@192.168.0.24:5060;ob>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Content-Type: application/sdp
Content-Length: 316
v=0
o=- 3768227747 3768227748 IN IP4 192.168.0.24
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4000 RTP/AVP 9 101
c=IN IP4 192.168.0.24
b=TIAS:64000
a=rtcp:4001 IN IP4 192.168.0.24
a=sendrecv
a=rtpmap:9 G722/8000
a=ssrc:1153692700 cname:3b0c6b47616e0ac7
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
--end msg--
=== Call state is now: INCOMING
...
--end msg--
=== Call state is now: CONNECTING
=== Call state is now: CONFIRMED
18:55:47.804 stream.c G722 codec used, remote samples per frame detected = 80
18:55:51.297 pjsua_core.c .RX 602 bytes Request msg BYE/cseq=5065495 (rdata0x7160129c) from UDP 192.168.0.234:5060:
...
--end msg--
18:55:51.298 pjsua_core.c .......TX 332 bytes Response msg 200/BYE/cseq=5065495 (tdta0x716128f4) to UDP 192.168.0.234:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.234;rport=5060;received=192.168.0.234;branch=z9hG4bKg61vKScF2eU2a
...
--end msg--
18:55:51.299 pjsua_media.c ......Call 0: deinitializing media..
18:55:51.300 pjsua_media.c ........Media stream call00:0 is destroyed
=== Call state is now: DISCONNECTED
当我挂断其他分机时，呼叫已断开