sip-lab 1.22.0 → 1.24.0

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

@@ -126,7 +126,6 @@ async function test() {
         },
     ], 2000)
 
-
     sip.call.reinvite(oc.id)
 
     await z.wait([
@@ -339,12 +338,59 @@ async function test() {
 
     await z.sleep(2000)
 
+    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.sleep(2000)
+
     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_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)