sip-lab 1.22.0 → 1.23.0

package/README.md

@@ -8,7 +8,7 @@ It uses pjproject for SIP and media processing.
 It permits to:
   - make audio calls using UDP, TCP and TLS transports
   - send/receive DTMF inband/RFC2833/INFO.
-  - play/record wav file on a call
+  - 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
@@ -17,8 +17,9 @@ It permits to:
 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
 

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

@@ -67,6 +67,7 @@ addon.call = {
   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": "1.22.0",
+  "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)
 

package/samples/text_to_speech.js

@@ -130,10 +130,29 @@ async function test() {
 	},
     ], 3000)
 
-    sip.call.start_speech_synth(oc.id, {voice: 'slt', text: 'Hello World.'})
-    sip.call.start_speech_synth(ic.id, {voice: 'kal', text: 'How are you?'})
+    sip.call.start_speech_synth(oc.id, {voice: 'slt', text: 'Hello World.', end_of_speech_event: true})
+    sip.call.start_speech_synth(ic.id, {voice: 'kal', text: 'How are you?', end_of_speech_event: true, no_loop: true})
 
-    await z.sleep(1500)
+    await z.wait([
+        {
+            event: 'end_of_speech',
+            call_id: ic.id,
+        },
+        {
+            event: 'end_of_speech',
+            call_id: oc.id,
+        },
+    ], 2000)
+
+    await z.wait([
+        {
+            event: 'end_of_speech',
+            call_id: oc.id,
+        },
+    ], 2000)
+
+    sip.call.stop_speech_synth(oc.id) // this is not actually necessary. It is used just to confirm the command works
+    sip.call.stop_speech_synth(ic.id) // this is not actually necessary. It is used just to confirm the command works
 
     sip.call.stop_record_wav(oc.id)
     sip.call.stop_record_wav(ic.id)

package/src/addon.cpp

@@ -704,6 +704,40 @@ Napi::Value call_stop_fax(const Napi::CallbackInfo &info) {
   return env.Null();
 }
 
+Napi::Value call_stop_speech_synth(const Napi::CallbackInfo &info) {
+  Napi::Env env = info.Env();
+
+  if (info.Length() != 2) {
+    Napi::Error::New(env, "Wrong number of arguments. Expected: call_id")
+        .ThrowAsJavaScriptException();
+    return env.Null();
+  }
+
+  if (!info[0].IsNumber()) {
+    Napi::TypeError::New(env, "call_id must be number.")
+        .ThrowAsJavaScriptException();
+    return env.Null();
+  }
+  int call_id = info[0].As<Napi::Number>().Int32Value();
+
+  if (!info[1].IsString()) {
+    Napi::TypeError::New(env, "params must be a JSON string.")
+        .ThrowAsJavaScriptException();
+    return env.Null();
+  }
+  const string json = info[1].As<Napi::String>().Utf8Value();
+
+  int res = pjw_call_stop_speech_synth(call_id, json.c_str());
+
+  if (res != 0) {
+    Napi::Error::New(env, pjw_get_error()).ThrowAsJavaScriptException();
+    return env.Null();
+  }
+
+  return env.Null();
+}
+
+
 Napi::Value call_get_stream_stat(const Napi::CallbackInfo &info) {
   Napi::Env env = info.Env();
 
@@ -1307,6 +1341,7 @@ Napi::Object init(Napi::Env env, Napi::Object exports) {
   exports.Set("call_stop_play_wav",
               Napi::Function::New(env, call_stop_play_wav));
   exports.Set("call_stop_fax", Napi::Function::New(env, call_stop_fax));
+  exports.Set("call_stop_speech_synth", Napi::Function::New(env, call_stop_speech_synth));
   exports.Set("call_get_stream_stat",
               Napi::Function::New(env, call_get_stream_stat));
   // exports.Set("call_refer", Napi::Function::New(env, call_refer));

package/src/event_templates.cpp

@@ -49,14 +49,9 @@ int make_evt_dtmf(char *dest, int size, long call_id, int digits_len,
 
 int make_evt_call_ended(char *dest, int size, long call_id, int sip_msg_len,
                         const char *sip_msg) {
-  printf("make_evt_call_ended sip_msg_len=%i sip_msg=%s\n", sip_msg_len,
+  printf("make_evt_call_ended sip_msg_len=%i sip_msg=%p\n", sip_msg_len,
          sip_msg);
-  if (!sip_msg || sip_msg == (char *)0xc000000000000) {
-    // received invalid pointer to sip_msg so do not add the message to the
-    // event
-    return snprintf(dest, size, "{\"event\": \"call_ended\", \"call_id\": %ld}",
-                    call_id);
-  } else if (sip_msg_len > 500 && sip_msg_len < 2000 && sip_msg) {
+  if (sip_msg_len > 500 && sip_msg_len < 2000 && sip_msg) {
     /* sip_msg_len sometimes show up as a large value like sip_msg_len=11560297
      * which seems to be a bug in pjsip */
     return snprintf(dest, size,
@@ -104,6 +99,18 @@ int make_evt_fax_result(char *dest, int size, long call_id, int result) {
       result);
 }
 
+int make_evt_end_of_file(char *dest, int size, long call_id) {
+  return snprintf(
+      dest, size,
+      "{\"event\": \"end_of_file\", \"call_id\": %ld}", call_id);
+}
+
+int make_evt_end_of_speech(char *dest, int size, long call_id) {
+  return snprintf(
+      dest, size,
+      "{\"event\": \"end_of_speech\", \"call_id\": %ld}", call_id);
+}
+
 int make_evt_tcp_msg(char *dest, int size, long call_id, const char *protocol, char *data, int data_len) {
   return snprintf(
       dest, size,