sip-lab 1.38.0 → 1.40.0

package/samples/opus.wideband.js

@@ -0,0 +1,247 @@
+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')
+
+var assert = require('assert')
+
+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", type: 'udp'})
+    t2 = sip.transport.create({address: "127.0.0.1", type: 'udp'})
+
+    console.log("t1", t1)
+    console.log("t2", t2)
+
+    console.log(sip.get_codecs())
+
+    sip.set_codecs("opus/48000/2:128")
+
+    sip.set_opus_config({
+      sample_rate: 48000,
+      channel_cnt: 2,
+      bit_rate: 16000,
+      packet_loss: 77,
+      complexity: 9,
+      cbr: true,
+      frm_ptime: 20
+    })
+
+    flags = 0
+
+    oc = sip.call.create(t1.id, {from_uri: 'sip:alice@test.com', to_uri: `sip:bob@${t2.address}:${t2.port}`})
+
+    await z.wait([
+        {
+            event: "incoming_call",
+            call_id: m.collect("call_id"),
+        },
+        {
+            event: 'response',
+            call_id: oc.id,
+            method: 'INVITE',
+            msg: sip_msg({
+                $rs: '100',
+                $rr: 'Trying',
+            }),
+        },
+    ], 1000)
+
+    ic = {
+        id: z.$call_id,
+        sip_call_id: z.$sip_call_id,
+    }
+
+    sip.call.respond(ic.id, {code: 200, reason: 'OK'})
+
+    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',
+                'hdr_content_type': 'application/sdp',
+                $rb: '!{_}a=sendrecv!{_}',
+            }),
+        },
+    ], 1000)
+
+    sip.call.start_record_wav(oc.id, {file: './oc.wav'})
+    sip.call.start_record_wav(ic.id, {file: './ic.wav'})
+
+    sip.call.start_play_wav(oc.id, {file: 'samples/artifacts/hello_good_morning.wav', end_of_file_event: true, no_loop: true})
+    sip.call.start_play_wav(ic.id, {file: 'samples/artifacts/hello_good_morning.wav', end_of_file_event: true, no_loop: true})
+
+    await z.wait([
+        {
+            event: 'end_of_file',
+            call_id: oc.id,
+        },
+        {
+            event: 'end_of_file',
+            call_id: ic.id,
+        },
+    ], 5000)
+
+    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(oc.id, false, 0)
+
+    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)
+
+    oc_stat = sip.call.get_stream_stat(oc.id, {media_id: 0})
+    ic_stat = sip.call.get_stream_stat(ic.id, {media_id: 0})
+
+    console.log(oc_stat)
+    console.log(ic_stat)
+
+    oc_stat = JSON.parse(oc_stat)
+    ic_stat = JSON.parse(ic_stat)
+
+    assert(oc_stat.CodecInfo == 'opus/48000/2')
+    assert(ic_stat.CodecInfo == 'opus/48000/2')
+
+    await z.sleep(100)
+
+    sip.call.send_dtmf(oc.id, {digits: '12', mode: 1})
+    sip.call.send_dtmf(ic.id, {digits: '21', mode: 1})
+
+    await z.sleep(1000)
+
+    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)
+
+    console.log("Success")
+
+    sip.stop()
+    process.exit(0)
+}
+
+test()
+.catch(e => {
+    console.error(e)
+    process.exit(1)
+})

package/samples/secure_websocket.js

@@ -0,0 +1,156 @@
+const sip = require('../index.js')
+const Zeq = require('@mayama/zeq')
+const m = require('data-matching')
+const sip_msg = require('sip-matching')
+
+var z = new Zeq()
+
+async function test() {
+    sip.dtmf_aggregation_on(500)
+
+    z.trap_events(sip.event_source, 'event', (evt) => {
+        return evt.args[0]
+    })
+
+    sip.set_codecs("pcmu/8000/1:128,pcma/8000/1:128")
+
+    console.log(sip.start((data) => { console.log(data) }))
+
+    // Create a WSS server transport (secure WebSocket listener)
+    const t2 = sip.transport.create({
+        address: "127.0.0.1",
+        port: 6062,
+        type: "wss",
+    })
+
+    // Create a WSS client transport connecting to our server
+    const t1 = sip.transport.create({
+        address: "127.0.0.1",
+        type: "wss",
+        ws_url: "wss://127.0.0.1:6062/sip",
+    })
+
+    console.log("t1", t1)
+    console.log("t2", t2)
+
+    // Make the call from t1 to t2 over Secure WebSocket
+    const oc = sip.call.create(t1.id, {
+        from_uri: 'sip:alice@test.com',
+        to_uri: 'sip:bob@127.0.0.1:6062',
+    })
+
+    // Wait for the call to arrive at t2 and 100 Trying response at t1
+    await z.wait([
+        {
+            event: "incoming_call",
+            call_id: m.collect("call_id"),
+            transport_id: t2.id,
+            msg: sip_msg({
+                $rU: 'bob',
+                $fU: 'alice',
+                $tU: 'bob',
+                $fd: 'test.com',
+            })
+        },
+        {
+            event: 'response',
+            call_id: oc.id,
+            method: 'INVITE',
+            msg: sip_msg({
+                $rs: '100',
+                $rr: 'Trying',
+            }),
+        },
+    ], 2000)
+
+    const ic = {
+        id: z.$call_id,
+        sip_call_id: z.$sip_call_id,
+    }
+
+    // Answer the call at t2 side
+    sip.call.respond(ic.id, {
+        code: 200,
+        reason: 'OK',
+    })
+
+    // Wait for 200 OK at t1 side and media setups
+    await z.wait([
+        {
+            event: 'response',
+            call_id: oc.id,
+            method: 'INVITE',
+            msg: sip_msg({
+                $rs: '200',
+                $rr: 'OK',
+            }),
+        },
+        {
+            event: 'media_update',
+            call_id: oc.id,
+            status: 'ok',
+        },
+        {
+            event: 'media_update',
+            call_id: ic.id,
+            status: 'ok',
+        },
+    ], 2000)
+
+    sip.call.start_inband_dtmf_detection(oc.id)
+    sip.call.start_inband_dtmf_detection(ic.id)
+
+    sip.call.send_dtmf(oc.id, {digits: '1234', mode: 1})
+    sip.call.send_dtmf(ic.id, {digits: '1234', mode: 1})
+
+    await z.wait([
+        {
+            event: 'dtmf',
+            call_id: ic.id,
+            digits: '1234',
+            mode: 1,
+            media_id: 0,
+        },
+        {
+            event: 'dtmf',
+            call_id: oc.id,
+            digits: '1234',
+            mode: 1,
+            media_id: 0,
+        },
+    ], 2000)
+
+    // Terminate the call from t1 side
+    sip.call.terminate(oc.id)
+
+    // Wait for call termination
+    await z.wait([
+        {
+            event: 'response',
+            call_id: oc.id,
+            method: 'BYE',
+            msg: sip_msg({
+                $rs: '200',
+                $rr: 'OK',
+            }),
+        },
+        {
+            event: 'call_ended',
+            call_id: oc.id,
+        },
+        {
+            event: 'call_ended',
+            call_id: ic.id,
+        },
+    ], 2000)
+
+    console.log("Secure WebSocket test successful")
+
+    sip.stop()
+    process.exit(0)
+}
+
+test().catch(e => {
+    console.error(e)
+    process.exit(1)
+})

package/samples/secure_websocket_opus_srtp_ice.js

@@ -0,0 +1,170 @@
+const sip = require('../index.js')
+const Zeq = require('@mayama/zeq')
+const m = require('data-matching')
+const sip_msg = require('sip-matching')
+const assert = require('assert')
+
+var z = new Zeq()
+
+async function test() {
+    sip.dtmf_aggregation_on(500)
+
+    z.trap_events(sip.event_source, 'event', (evt) => {
+        return evt.args[0]
+    })
+
+    sip.set_codecs("opus/48000/2:128,pcmu/8000/1:128")
+
+    console.log(sip.start((data) => { console.log(data) }))
+
+    // Create a WSS server transport (secure WebSocket listener)
+    const t2 = sip.transport.create({
+        address: "127.0.0.1",
+        port: 6062,
+        type: "wss",
+    })
+
+    // Create a WSS client transport connecting to our server
+    const t1 = sip.transport.create({
+        address: "127.0.0.1",
+        type: "wss",
+        ws_url: "wss://127.0.0.1:6062/sip",
+    })
+
+    console.log("t1", t1)
+    console.log("t2", t2)
+
+    // Make the call from t1 to t2 over Secure WebSocket
+    // Use OPUS codec with SRTP and ICE
+    const oc = sip.call.create(t1.id, {
+        from_uri: 'sip:alice@test.com',
+        to_uri: 'sip:bob@127.0.0.1:6062',
+        media: [{type: "audio", secure: true, ice: true}],
+    })
+
+    // Wait for the call to arrive at t2 and 100 Trying response at t1
+    await z.wait([
+        {
+            event: "incoming_call",
+            call_id: m.collect("call_id"),
+            transport_id: t2.id,
+            msg: sip_msg({
+                $rU: 'bob',
+                $fU: 'alice',
+                $tU: 'bob',
+                $fd: 'test.com',
+            })
+        },
+        {
+            event: 'response',
+            call_id: oc.id,
+            method: 'INVITE',
+            msg: sip_msg({
+                $rs: '100',
+                $rr: 'Trying',
+            }),
+        },
+    ], 2000)
+
+    const ic = {
+        id: z.$call_id,
+        sip_call_id: z.$sip_call_id,
+    }
+
+    // Answer the call at t2 side with matching media config
+    sip.call.respond(ic.id, {
+        code: 200,
+        reason: 'OK',
+        media: [{type: "audio", secure: true, ice: true}],
+    })
+
+    // Wait for 200 OK at t1 side and media setups
+    await z.wait([
+        {
+            event: 'response',
+            call_id: oc.id,
+            method: 'INVITE',
+            msg: sip_msg({
+                $rs: '200',
+                $rr: 'OK',
+            }),
+        },
+        {
+            event: 'media_update',
+            call_id: oc.id,
+            status: 'ok',
+        },
+        {
+            event: 'media_update',
+            call_id: ic.id,
+            status: 'ok',
+        },
+    ], 5000)
+
+    sip.call.start_inband_dtmf_detection(oc.id)
+    sip.call.start_inband_dtmf_detection(ic.id)
+
+    // using 1234 fails frequently as we get things like '12334'
+    sip.call.send_dtmf(oc.id, {digits: '12', mode: 1})
+    sip.call.send_dtmf(ic.id, {digits: '12', mode: 1})
+
+    await z.wait([
+        {
+            event: 'dtmf',
+            call_id: ic.id,
+            digits: '12',
+            mode: 1,
+            media_id: 0,
+        },
+        {
+            event: 'dtmf',
+            call_id: oc.id,
+            digits: '12',
+            mode: 1,
+            media_id: 0,
+        },
+    ], 2000)
+
+    stat1 = JSON.parse(sip.call.get_stream_stat(oc.id, {media_id: 0}))
+    stat2 = JSON.parse(sip.call.get_stream_stat(ic.id, {media_id: 0}))
+
+    console.log("stat1", stat1)
+    console.log("stat2", stat2)
+
+    assert(stat1.CodecInfo == "opus/8000/1")
+    assert(stat2.CodecInfo == "opus/8000/1")
+
+    // Terminate the call from t1 side
+    sip.call.terminate(oc.id)
+
+    // Wait for call termination
+    await z.wait([
+        {
+            event: 'response',
+            call_id: oc.id,
+            method: 'BYE',
+            msg: sip_msg({
+                $rs: '200',
+                $rr: 'OK',
+            }),
+        },
+        {
+            event: 'call_ended',
+            call_id: oc.id,
+        },
+        {
+            event: 'call_ended',
+            call_id: ic.id,
+        },
+    ], 2000)
+
+    console.log("Secure WebSocket + OPUS + SRTP + ICE test successful")
+
+    sip.stop()
+    process.exit(0)
+}
+
+test().catch(e => {
+    console.error(e)
+    process.exit(1)
+})

package/samples/simple.js

@@ -10,6 +10,9 @@ var z = new Zeq()
 
 
 async function test() {
+    // here we set DTMF aggregation mode to 500 ms
+    sip.dtmf_aggregation_on(500)
+
     // here we set our Zeq instance to trap events generated by sip-lab event_source
     z.trap_events(sip.event_source, 'event', (evt) => {
         var e = evt.args[0]
@@ -147,6 +150,29 @@ async function test() {
         },
     ], 1000)
 
+    sip.call.start_inband_dtmf_detection(oc.id)
+    sip.call.start_inband_dtmf_detection(ic.id)
+
+    sip.call.send_dtmf(oc.id, {digits: '1234', mode: 1})
+    sip.call.send_dtmf(ic.id, {digits: '1234', mode: 1})
+
+    await z.wait([
+        {
+            event: 'dtmf',
+            call_id: ic.id,
+            digits: '1234',
+            mode: 1,
+            media_id: 0,
+        },
+        {
+            event: 'dtmf',
+            call_id: oc.id,
+            digits: '1234',
+            mode: 1,
+            media_id: 0,
+        },
+    ], 2000)
+
     // now we terminate the call from t1 side
     sip.call.terminate(oc.id)