sip-lab 1.21.0 → 1.23.0

package/README.md

@@ -7,15 +7,19 @@ It uses pjproject for SIP and media processing.
 
 It permits to:
   - make audio calls using UDP, TCP and TLS transports
-  - send and receive DTMF inband/RFC2833/INFO.
-  - play/record wav file on a call
+  - send/receive DTMF inband/RFC2833/INFO.
+  - play/record audio on a call from/to a wav file
   - send/receive fax (T.30 only)
   - send/receive MRCPv2 messages (TCP only, no TLS)
+  - send/receive audio using SRTP
+  - generate speech from text into a call (using flite)
 
 TODO:
   - add support for video playing/recording from/to file
+  - add support for speech recognition using pocketsphinx
+  - add support for speech synth/recog via websocket server
   - add support for T.38 fax
-  - add support for WebSocket
+  - add support for SIP over WebSocket
   - add support for WebRTC
   - add support for MSRP
 
@@ -26,7 +30,7 @@ It is distributed with prebuild binaries for node.js 15.0.0 and above (but built
 
 To install it, first install build dependencies:
 ```
-apt install build-essential automake autoconf libtool libspeex-dev libopus-dev libsdl2-dev libavdevice-dev libswscale-dev libv4l-dev libopencore-amrnb-dev libopencore-amrwb-dev libvo-amrwbenc-dev libvo-amrwbenc-dev libboost-dev libtiff-dev libpcap-dev libssl-dev uuid-dev cmake
+apt install build-essential automake autoconf libtool libspeex-dev libopus-dev libsdl2-dev libavdevice-dev libswscale-dev libv4l-dev libopencore-amrnb-dev libopencore-amrwb-dev libvo-amrwbenc-dev libvo-amrwbenc-dev libboost-dev libtiff-dev libpcap-dev libssl-dev uuid-dev flite-dev cmake
 ```
 
 Then install sip-lab (local build of the addon might be triggered here if this is not Debian 11):

package/binding.gyp

@@ -80,6 +80,12 @@
                 '-lswscale',
                 '-lavutil',
                 '-lspeex',
+		'-lflite',
+		'-lflite_cmu_us_awb',
+		'-lflite_cmu_us_kal',
+		'-lflite_cmu_us_rms',
+		'-lflite_cmu_us_slt',
+		'-lflite_cmu_us_kal16',
                 '-l srtp-x86_64-unknown-linux-gnu',
               ],
           },
@@ -109,6 +115,7 @@
         'src/addon.cpp',
         'src/pjmedia/src/pjmedia/dtmfdet.c',
         'src/pjmedia/src/pjmedia/fax_port.c',
+        'src/pjmedia/src/pjmedia/flite_port.c',
       ],
     },
   ],

package/build_deps.sh

@@ -55,7 +55,8 @@ then
     #git checkout 33a3c9e0a5eb84426edef05a9aa98af17d8011c3 # required for bcg729
     #git checkout 797088ed133c98492519b7d042b75735f6f9388c # updated as part of #21
     #git checkout 651df5b50129b7c5a5feec8336dda4468d53d2b0 # updated to latest to see of crash issues improve
-    git checkout 043926a5846963a2c99378e8daa495230923eaab # update to try to solve ##49 (but issue remains)
+    #git checkout 043926a5846963a2c99378e8daa495230923eaab # updated to try to solve #49 (but issue remains)
+    git checkout c36802585ddefb3ca477d1f6d773d179510c5412 # updated to try to solve #83 (but issue remains)
 
     cat > user.mak <<EOF
     export CFLAGS += -fPIC -g

package/index.js

@@ -66,6 +66,8 @@ addon.call = {
   stop_play_wav: (c_id, params) => { return addon.call_stop_play_wav(c_id, JSON.stringify(params ? params : {})) },
   start_fax: (c_id, params) => { return addon.call_start_fax(c_id, JSON.stringify(params)) },
   stop_fax: (c_id, params) => { return addon.call_stop_fax(c_id, JSON.stringify(params ? params : {})) },
+  start_speech_synth: (c_id, params) => { return addon.call_start_speech_synth(c_id, JSON.stringify(params)) },
+  stop_speech_synth: (c_id, params) => { return addon.call_stop_speech_synth(c_id, JSON.stringify(params ? params : {})) },
   get_stream_stat: (c_id, params) => { return addon.call_get_stream_stat(c_id, JSON.stringify(params ? params : {})) },
   //refer: (c_id, params) => { return addon.call_refer(c_id, JSON.stringify(params)) },
   get_info: addon.call_get_info,

package/package.json

@@ -1,6 +1,6 @@
 {
   "name": "sip-lab",
-  "version": "1.21.0",
+  "version": "1.23.0",
   "description": "",
   "main": "index.js",
   "engines": {

package/samples/start_play_wav_with_end_of_file_event.js

@@ -0,0 +1,269 @@
+var sip = require ('../index.js')
+var Zeq = require('@mayama/zeq')
+var z = new Zeq()
+var m = require('data-matching')
+var sip_msg = require('sip-matching')
+var sdp = require('sdp-matching')
+
+async function test() {
+    //sip.set_log_level(6)
+    sip.dtmf_aggregation_on(500)
+
+    z.trap_events(sip.event_source, 'event', (evt) => {
+        var e = evt.args[0]
+        return e
+    })
+
+    console.log(sip.start((data) => { console.log(data)} ))
+
+    t1 = sip.transport.create({address: "127.0.0.1"})
+    t2 = sip.transport.create({address: "127.0.0.1"})
+
+    console.log("t1", t1)
+    console.log("t2", t2)
+
+    oc = sip.call.create(t1.id, {
+        from_uri: '"abc"<sip:alice@test.com>',
+        to_uri: `sip:bob@${t2.address}:${t2.port}`,
+        headers: {
+            'X-MyHeader1': 'abc',
+            'X-MyHeader2': 'def',
+        },
+    })
+
+    await z.wait([
+        {
+            event: "incoming_call",
+            call_id: m.collect("call_id"),
+            msg: sip_msg({
+                $rm: 'INVITE',
+                $fU: 'alice',
+                $fd: 'test.com',
+                $tU: 'bob',
+                '$hdr(X-MyHeader1)': 'abc',
+                'hdr_x_myheader2': 'def',
+            }),
+        },
+        {
+            event: 'response',
+            call_id: oc.id,
+            method: 'INVITE',
+            msg: sip_msg({
+                $rs: '100',
+                $rr: 'Trying',
+                '$(hdrcnt(via))': 1,
+                'hdr_call_id': m.collect('sip_call_id'),
+                $fU: 'alice',
+                $fd: 'test.com',
+                $tU: 'bob',
+                '$hdr(l)': '0',
+            }),
+        },
+    ], 1000)
+
+    ic = {
+        id: z.store.call_id,
+        sip_call_id: z.store.sip_call_id,
+    }
+
+    sip.call.respond(ic.id, {
+        code: 200,
+        reason:'OK',
+        headers: {
+            'X-MyHeader3': 'ghi',
+            'X-MyHeader4': 'jkl',
+        },
+    })
+
+    await z.wait([
+        {
+            event: 'media_update',
+            call_id: oc.id,
+            status: 'ok',
+        },
+        {
+            event: 'media_update',
+            call_id: ic.id,
+            status: 'ok',
+        },
+        {
+            event: 'response',
+            call_id: oc.id,
+            method: 'INVITE',
+            msg: sip_msg({
+                $rs: '200',
+                $rr: 'OK',
+                '$(hdrcnt(v))': 1,
+                $fU: 'alice',
+                $fd: 'test.com',
+                $tU: 'bob',
+                '$hdr(content-type)': 'application/sdp',
+                $rb: '!{_}a=sendrecv',
+                '$hdr(X-MyHeader3)': 'ghi',
+                '$hdr(X-MyHeader4)': 'jkl',
+            }),
+        },
+    ], 1000)
+
+    sip.call.start_record_wav(oc.id, {file: './oc.wav'})
+    sip.call.start_record_wav(ic.id, {file: './ic.wav'})
+
+    await z.sleep(100)
+
+    sip.call.start_play_wav(oc.id, {file: 'samples/artifacts/yosemitesam.wav', end_of_file_event: true})
+    sip.call.start_play_wav(ic.id, {file: 'samples/artifacts/yosemitesam.wav', end_of_file_event: true})
+
+    await z.sleep(500)
+
+    sip.call.reinvite(oc.id)
+
+    await z.wait([
+        {
+            event: 'reinvite',
+            call_id: ic.id
+        },
+    ], 1000)
+
+    sip.call.respond(ic.id, {code: 200, reason: 'OK'})
+
+    await z.wait([
+        {
+            event: 'response',
+            call_id: oc.id,
+            method: 'INVITE',
+            msg: sip_msg({
+                $rs: '100',
+            }),
+        },
+        {
+            event: 'response',
+            call_id: oc.id,
+            method: 'INVITE',
+            msg: sip_msg({
+                $rs: '200',
+                $rr: 'OK',
+                $rb: '!{_}a=sendrecv',
+            }),
+        },
+        {
+            event: 'media_update',
+            call_id: oc.id,
+            status: 'ok',
+        },
+        {
+            event: 'media_update',
+            call_id: ic.id,
+            status: 'ok',
+        },
+    ], 500)
+
+    await z.wait([
+        {
+            event: 'end_of_file',
+            call_id: ic.id,
+        },
+        {
+            event: 'end_of_file',
+            call_id: oc.id,
+        },
+    ], 2000)
+
+
+    sip.call.reinvite(ic.id)
+
+    await z.wait([
+        {
+            event: 'reinvite',
+            call_id: oc.id
+        },
+    ], 1000)
+
+    sip.call.respond(oc.id, {code: 200, reason: 'OK'})
+
+    await z.wait([
+        {
+            event: 'response',
+            call_id: ic.id,
+            method: 'INVITE',
+            msg: sip_msg({
+                $rs: '100',
+            }),
+        },
+        {
+            event: 'response',
+            call_id: ic.id,
+            method: 'INVITE',
+            msg: sip_msg({
+                $rs: '200',
+                $rr: 'OK',
+                $rb: '!{_}a=sendrecv',
+            }),
+        },
+        {
+            event: 'media_update',
+            call_id: oc.id,
+            status: 'ok',
+        },
+        {
+            event: 'media_update',
+            call_id: ic.id,
+            status: 'ok',
+        },
+    ], 500)
+
+    await z.wait([
+        {
+            event: 'end_of_file',
+            call_id: ic.id,
+        },
+        {
+            event: 'end_of_file',
+            call_id: oc.id,
+        },
+    ], 5000)
+
+    stat1 = sip.call.get_stream_stat(oc.id, {media_id: 0})
+    stat2 = sip.call.get_stream_stat(ic.id, {media_id: 0})
+
+    console.log("stat1", stat1)
+    console.log("stat2", stat2)
+
+    sip.call.stop_record_wav(oc.id)
+    sip.call.stop_record_wav(ic.id)
+
+    sip.call.terminate(oc.id)
+
+    await z.wait([
+        {
+            event: 'call_ended',
+            call_id: oc.id,
+        },
+        {
+            event: 'call_ended',
+            call_id: ic.id,
+        },
+        {
+            event: 'response',
+            call_id: oc.id,
+            method: 'BYE',
+            msg: sip_msg({
+                $rs: '200',
+                $rr: 'OK',
+            }),
+        },
+    ], 1000)
+
+    await z.sleep(1000)
+
+    console.log("Success")
+
+    sip.stop()
+}
+
+
+test()
+.catch(e => {
+    console.error(e)
+    process.exit(1)
+})
+

package/samples/start_play_wav_with_no_loop.js

@@ -0,0 +1,257 @@
+var sip = require ('../index.js')
+var Zeq = require('@mayama/zeq')
+var z = new Zeq()
+var m = require('data-matching')
+var sip_msg = require('sip-matching')
+var sdp = require('sdp-matching')
+
+async function test() {
+    //sip.set_log_level(6)
+    sip.dtmf_aggregation_on(500)
+
+    z.trap_events(sip.event_source, 'event', (evt) => {
+        var e = evt.args[0]
+        return e
+    })
+
+    console.log(sip.start((data) => { console.log(data)} ))
+
+    t1 = sip.transport.create({address: "127.0.0.1"})
+    t2 = sip.transport.create({address: "127.0.0.1"})
+
+    console.log("t1", t1)
+    console.log("t2", t2)
+
+    oc = sip.call.create(t1.id, {
+        from_uri: '"abc"<sip:alice@test.com>',
+        to_uri: `sip:bob@${t2.address}:${t2.port}`,
+        headers: {
+            'X-MyHeader1': 'abc',
+            'X-MyHeader2': 'def',
+        },
+    })
+
+    await z.wait([
+        {
+            event: "incoming_call",
+            call_id: m.collect("call_id"),
+            msg: sip_msg({
+                $rm: 'INVITE',
+                $fU: 'alice',
+                $fd: 'test.com',
+                $tU: 'bob',
+                '$hdr(X-MyHeader1)': 'abc',
+                'hdr_x_myheader2': 'def',
+            }),
+        },
+        {
+            event: 'response',
+            call_id: oc.id,
+            method: 'INVITE',
+            msg: sip_msg({
+                $rs: '100',
+                $rr: 'Trying',
+                '$(hdrcnt(via))': 1,
+                'hdr_call_id': m.collect('sip_call_id'),
+                $fU: 'alice',
+                $fd: 'test.com',
+                $tU: 'bob',
+                '$hdr(l)': '0',
+            }),
+        },
+    ], 1000)
+
+    ic = {
+        id: z.store.call_id,
+        sip_call_id: z.store.sip_call_id,
+    }
+
+    sip.call.respond(ic.id, {
+        code: 200,
+        reason:'OK',
+        headers: {
+            'X-MyHeader3': 'ghi',
+            'X-MyHeader4': 'jkl',
+        },
+    })
+
+    await z.wait([
+        {
+            event: 'media_update',
+            call_id: oc.id,
+            status: 'ok',
+        },
+        {
+            event: 'media_update',
+            call_id: ic.id,
+            status: 'ok',
+        },
+        {
+            event: 'response',
+            call_id: oc.id,
+            method: 'INVITE',
+            msg: sip_msg({
+                $rs: '200',
+                $rr: 'OK',
+                '$(hdrcnt(v))': 1,
+                $fU: 'alice',
+                $fd: 'test.com',
+                $tU: 'bob',
+                '$hdr(content-type)': 'application/sdp',
+                $rb: '!{_}a=sendrecv',
+                '$hdr(X-MyHeader3)': 'ghi',
+                '$hdr(X-MyHeader4)': 'jkl',
+            }),
+        },
+    ], 1000)
+
+    sip.call.start_record_wav(oc.id, {file: './oc.wav'})
+    sip.call.start_record_wav(ic.id, {file: './ic.wav'})
+
+    await z.sleep(100)
+
+    sip.call.start_play_wav(oc.id, {file: 'samples/artifacts/yosemitesam.wav', end_of_file_event: true, no_loop: true})
+    sip.call.start_play_wav(ic.id, {file: 'samples/artifacts/yosemitesam.wav', end_of_file_event: true, no_loop: true})
+
+    sip.call.reinvite(oc.id)
+
+    await z.wait([
+        {
+            event: 'reinvite',
+            call_id: ic.id
+        },
+    ], 1000)
+
+    sip.call.respond(ic.id, {code: 200, reason: 'OK'})
+
+    await z.wait([
+        {
+            event: 'response',
+            call_id: oc.id,
+            method: 'INVITE',
+            msg: sip_msg({
+                $rs: '100',
+            }),
+        },
+        {
+            event: 'response',
+            call_id: oc.id,
+            method: 'INVITE',
+            msg: sip_msg({
+                $rs: '200',
+                $rr: 'OK',
+                $rb: '!{_}a=sendrecv',
+            }),
+        },
+        {
+            event: 'media_update',
+            call_id: oc.id,
+            status: 'ok',
+        },
+        {
+            event: 'media_update',
+            call_id: ic.id,
+            status: 'ok',
+        },
+    ], 500)
+
+    sip.call.reinvite(ic.id)
+
+    await z.wait([
+        {
+            event: 'reinvite',
+            call_id: oc.id
+        },
+    ], 1000)
+
+    sip.call.respond(oc.id, {code: 200, reason: 'OK'})
+
+    await z.wait([
+        {
+            event: 'response',
+            call_id: ic.id,
+            method: 'INVITE',
+            msg: sip_msg({
+                $rs: '100',
+            }),
+        },
+        {
+            event: 'response',
+            call_id: ic.id,
+            method: 'INVITE',
+            msg: sip_msg({
+                $rs: '200',
+                $rr: 'OK',
+                $rb: '!{_}a=sendrecv',
+            }),
+        },
+        {
+            event: 'media_update',
+            call_id: oc.id,
+            status: 'ok',
+        },
+        {
+            event: 'media_update',
+            call_id: ic.id,
+            status: 'ok',
+        },
+    ], 500)
+
+    await z.wait([
+        {
+            event: 'end_of_file',
+            call_id: ic.id,
+        },
+        {
+            event: 'end_of_file',
+            call_id: oc.id,
+        },
+    ], 3000)
+
+    await z.sleep(3000) // we should not receive end_of_file events again
+
+    stat1 = sip.call.get_stream_stat(oc.id, {media_id: 0})
+    stat2 = sip.call.get_stream_stat(ic.id, {media_id: 0})
+
+    console.log("stat1", stat1)
+    console.log("stat2", stat2)
+
+    sip.call.stop_record_wav(oc.id)
+    sip.call.stop_record_wav(ic.id)
+
+    sip.call.terminate(oc.id)
+
+    await z.wait([
+        {
+            event: 'call_ended',
+            call_id: oc.id,
+        },
+        {
+            event: 'call_ended',
+            call_id: ic.id,
+        },
+        {
+            event: 'response',
+            call_id: oc.id,
+            method: 'BYE',
+            msg: sip_msg({
+                $rs: '200',
+                $rr: 'OK',
+            }),
+        },
+    ], 1000)
+
+    await z.sleep(1000)
+
+    console.log("Success")
+
+    sip.stop()
+}
+
+
+test()
+.catch(e => {
+    console.error(e)
+    process.exit(1)
+})
+

package/samples/tcp_and_extra_headers.js

@@ -345,6 +345,9 @@ async function test() {
     console.log("stat1", stat1)
     console.log("stat2", stat2)
 
+    sip.call.stop_play_wav(oc.id) // this is not really necessary. We are just confirming it works
+    sip.call.stop_play_wav(ic.id) // this is not really necessary. We are just confirming it works
+
     sip.call.stop_record_wav(oc.id)
     sip.call.stop_record_wav(ic.id)