sip-lab 1.28.0 → 1.28.2
This diff represents the content of publicly available package versions that have been released to one of the supported registries. The information contained in this diff is provided for informational purposes only and reflects changes between package versions as they appear in their respective public registries.
package/README.md
CHANGED
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@@ -40,7 +40,7 @@ npm install sip-lab
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Then run some sample script from subfolder samples:
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```
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-
node samples/simple.js
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+
node node_modules/sip-lab/samples/simple.js
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```
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The above script has detailed comments.
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@@ -56,7 +56,7 @@ There are additional samples scripts in folder samples_extra but they require [w
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To run ws_speech_server, do this:
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```
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-
https://github.com/MayamaTakeshi/ws_speech_server
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git clone https://github.com/MayamaTakeshi/ws_speech_server
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cd ws_speech_server
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npm i
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npm run build
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@@ -67,11 +67,11 @@ node src/App.bs.js
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Then you should be able to test with dtmf language:
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```
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node samples_extra/ws_speech_server.dtmf.js
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node node_modules/sip-lab/samples_extra/ws_speech_server.dtmf.js
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```
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or with google speech service:
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```
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node samples_extra/ws_speech_server.google.js
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node node_modules/sip-lab/samples_extra/ws_speech_server.google.js
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```
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package/package.json
CHANGED
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@@ -1,6 +1,6 @@
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{
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"name": "sip-lab",
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-
"version": "1.28.
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+
"version": "1.28.2",
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"description": "",
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"main": "index.js",
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"engines": {
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@@ -44,6 +44,7 @@
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"README.md",
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"DEV.md",
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"samples",
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"samples_extra",
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"prebuilds",
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"pocketsphinx",
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"runtests"
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@@ -0,0 +1,194 @@
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var sip = require ('../index.js')
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var Zeq = require('@mayama/zeq')
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var z = new Zeq()
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var m = require('data-matching')
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var sip_msg = require('sip-matching')
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var sdp = require('sdp-matching')
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async function test() {
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//sip.set_log_level(6)
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sip.dtmf_aggregation_on(500)
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sip.set_codecs("PCMU/8000/1:128")
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z.trap_events(sip.event_source, 'event', (evt) => {
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var e = evt.args[0]
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return e
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})
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console.log(sip.start((data) => { console.log(data)} ))
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t1 = sip.transport.create({address: "127.0.0.1", type: 'tcp'})
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t2 = sip.transport.create({address: "127.0.0.1", type: 'tcp'})
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console.log("t1", t1)
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console.log("t2", t2)
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oc = sip.call.create(t1.id, {
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from_uri: '"abc"<sip:alice@test.com>',
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to_uri: `sip:bob@${t2.address}:${t2.port}`,
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headers: {
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'X-MyHeader1': 'abc',
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'X-MyHeader2': 'def',
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},
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})
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await z.wait([
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{
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event: "incoming_call",
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call_id: m.collect("call_id"),
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msg: sip_msg({
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$rm: 'INVITE',
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$fU: 'alice',
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$fd: 'test.com',
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$tU: 'bob',
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'$hdr(X-MyHeader1)': 'abc',
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'hdr_x_myheader2': 'def',
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}),
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},
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{
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event: 'response',
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call_id: oc.id,
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method: 'INVITE',
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msg: sip_msg({
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$rs: '100',
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$rr: 'Trying',
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'$(hdrcnt(via))': 1,
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'hdr_call_id': m.collect('sip_call_id'),
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$fU: 'alice',
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$fd: 'test.com',
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$tU: 'bob',
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'$hdr(l)': '0',
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}),
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},
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], 1000)
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ic = {
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id: z.store.call_id,
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sip_call_id: z.store.sip_call_id,
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}
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sip.call.respond(ic.id, {
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code: 200,
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reason:'OK',
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headers: {
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'X-MyHeader3': 'ghi',
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'X-MyHeader4': 'jkl',
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},
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})
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await z.wait([
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{
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event: 'media_update',
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call_id: oc.id,
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status: 'ok',
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},
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{
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event: 'media_update',
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call_id: ic.id,
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status: 'ok',
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},
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{
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event: 'response',
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call_id: oc.id,
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method: 'INVITE',
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msg: sip_msg({
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$rs: '200',
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$rr: 'OK',
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'$(hdrcnt(v))': 1,
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$fU: 'alice',
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$fd: 'test.com',
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$tU: 'bob',
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'$hdr(content-type)': 'application/sdp',
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$rb: '!{_}a=sendrecv',
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'$hdr(X-MyHeader3)': 'ghi',
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'$hdr(X-MyHeader4)': 'jkl',
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}),
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},
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], 1000)
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sip.call.start_record_wav(oc.id, {file: './oc.wav'})
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sip.call.start_record_wav(ic.id, {file: './ic.wav'})
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sip.call.start_speech_synth(oc.id, {server_url: 'ws://0.0.0.0:8080', engine: 'dtmf-gen', voice: 'dtmf', language: 'dtmf', text: 'abcd', times: 1})
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sip.call.start_speech_synth(ic.id, {server_url: 'ws://0.0.0.0:8080', engine: 'dtmf-gen', voice: 'dtmf', language: 'dtmf', text: 'dcba', times: 1})
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sip.call.start_speech_recog(oc.id, {server_url: 'ws://0.0.0.0:8080', engine: 'dtmf-det', language: 'dtmf'})
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sip.call.start_speech_recog(ic.id, {server_url: 'ws://0.0.0.0:8080', engine: 'dtmf-det', language: 'dtmf'})
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await z.wait([
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{
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event: 'speech_synth_complete',
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call_id: ic.id,
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},
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{
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event: 'speech_synth_complete',
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call_id: oc.id,
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},
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{
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event: 'dtmf',
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call_id: oc.id,
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digits: 'dcba',
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mode: 1,
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media_id: 0
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},
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{
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event: 'dtmf',
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call_id: ic.id,
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digits: 'abcd',
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mode: 1,
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media_id: 0
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},
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{
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event: 'speech',
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call_id: oc.id,
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transcript: 'dcba'
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},
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{
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event: 'speech',
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call_id: ic.id,
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transcript: 'abcd'
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},
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], 3000)
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await z.sleep(1000)
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sip.call.stop_record_wav(oc.id)
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sip.call.stop_record_wav(ic.id)
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sip.call.terminate(oc.id)
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await z.wait([
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{
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event: 'call_ended',
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call_id: oc.id,
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},
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{
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event: 'call_ended',
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call_id: ic.id,
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},
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{
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event: 'response',
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call_id: oc.id,
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method: 'BYE',
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msg: sip_msg({
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$rs: '200',
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$rr: 'OK',
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}),
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},
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], 1000)
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await z.sleep(100)
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console.log("Success")
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sip.stop()
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}
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test()
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.catch(e => {
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console.error(e)
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process.exit(1)
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})
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@@ -0,0 +1,190 @@
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var sip = require ('../index.js')
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var Zeq = require('@mayama/zeq')
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var z = new Zeq()
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var m = require('data-matching')
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var sip_msg = require('sip-matching')
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var sdp = require('sdp-matching')
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async function test() {
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//sip.set_log_level(6)
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sip.dtmf_aggregation_on(500)
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sip.set_codecs("PCMU/8000/1:128")
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z.trap_events(sip.event_source, 'event', (evt) => {
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var e = evt.args[0]
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return e
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})
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console.log(sip.start((data) => { console.log(data)} ))
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t1 = sip.transport.create({address: "127.0.0.1", type: 'tcp'})
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t2 = sip.transport.create({address: "127.0.0.1", type: 'tcp'})
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console.log("t1", t1)
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console.log("t2", t2)
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oc = sip.call.create(t1.id, {
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from_uri: '"abc"<sip:alice@test.com>',
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to_uri: `sip:bob@${t2.address}:${t2.port}`,
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headers: {
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'X-MyHeader1': 'abc',
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'X-MyHeader2': 'def',
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},
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})
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await z.wait([
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{
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event: "incoming_call",
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call_id: m.collect("call_id"),
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msg: sip_msg({
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$rm: 'INVITE',
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$fU: 'alice',
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$fd: 'test.com',
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$tU: 'bob',
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'$hdr(X-MyHeader1)': 'abc',
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'hdr_x_myheader2': 'def',
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}),
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},
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{
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event: 'response',
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call_id: oc.id,
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method: 'INVITE',
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msg: sip_msg({
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$rs: '100',
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$rr: 'Trying',
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'$(hdrcnt(via))': 1,
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'hdr_call_id': m.collect('sip_call_id'),
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$fU: 'alice',
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$fd: 'test.com',
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$tU: 'bob',
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'$hdr(l)': '0',
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}),
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},
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], 1000)
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ic = {
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id: z.store.call_id,
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sip_call_id: z.store.sip_call_id,
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}
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sip.call.respond(ic.id, {
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code: 200,
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reason:'OK',
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headers: {
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'X-MyHeader3': 'ghi',
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'X-MyHeader4': 'jkl',
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},
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})
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80
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await z.wait([
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{
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event: 'media_update',
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call_id: oc.id,
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status: 'ok',
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},
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{
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event: 'media_update',
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call_id: ic.id,
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status: 'ok',
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},
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{
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event: 'response',
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call_id: oc.id,
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method: 'INVITE',
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msg: sip_msg({
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$rs: '200',
|
|
97
|
+
$rr: 'OK',
|
|
98
|
+
'$(hdrcnt(v))': 1,
|
|
99
|
+
$fU: 'alice',
|
|
100
|
+
$fd: 'test.com',
|
|
101
|
+
$tU: 'bob',
|
|
102
|
+
'$hdr(content-type)': 'application/sdp',
|
|
103
|
+
$rb: '!{_}a=sendrecv',
|
|
104
|
+
'$hdr(X-MyHeader3)': 'ghi',
|
|
105
|
+
'$hdr(X-MyHeader4)': 'jkl',
|
|
106
|
+
}),
|
|
107
|
+
},
|
|
108
|
+
], 1000)
|
|
109
|
+
|
|
110
|
+
await z.sleep(200)
|
|
111
|
+
|
|
112
|
+
sip.call.start_record_wav(oc.id, {file: './oc.wav'})
|
|
113
|
+
sip.call.start_record_wav(ic.id, {file: './ic.wav'})
|
|
114
|
+
|
|
115
|
+
sip.call.start_speech_synth(oc.id, {server_url: 'ws://0.0.0.0:8080', engine: 'gss', voice: 'en-US-Standard-E', language: 'en-US', text: 'hello world', times: 1})
|
|
116
|
+
sip.call.start_speech_synth(ic.id, {server_url: 'ws://0.0.0.0:8080', engine: 'gss', voice: 'en-US-Standard-F', language: 'en-US', text: '<speak>Good morning<break time="2s"/>Good Afternoon</speak>', times: 1})
|
|
117
|
+
|
|
118
|
+
sip.call.start_speech_recog(oc.id, {server_url: 'ws://0.0.0.0:8080', engine: 'gsr', language: 'en-US'})
|
|
119
|
+
sip.call.start_speech_recog(ic.id, {server_url: 'ws://0.0.0.0:8080', engine: 'gsr', language: 'en-US'})
|
|
120
|
+
|
|
121
|
+
await z.wait([
|
|
122
|
+
{
|
|
123
|
+
event: 'speech_synth_complete',
|
|
124
|
+
call_id: ic.id,
|
|
125
|
+
},
|
|
126
|
+
{
|
|
127
|
+
event: 'speech_synth_complete',
|
|
128
|
+
call_id: oc.id,
|
|
129
|
+
},
|
|
130
|
+
{
|
|
131
|
+
event: 'speech',
|
|
132
|
+
call_id: oc.id,
|
|
133
|
+
transcript: 'good morning',
|
|
134
|
+
},
|
|
135
|
+
{
|
|
136
|
+
event: 'speech',
|
|
137
|
+
call_id: ic.id,
|
|
138
|
+
transcript: 'hello world',
|
|
139
|
+
},
|
|
140
|
+
], 4000)
|
|
141
|
+
|
|
142
|
+
await z.wait([
|
|
143
|
+
{
|
|
144
|
+
event: 'speech',
|
|
145
|
+
call_id: oc.id,
|
|
146
|
+
transcript: ' good afternoon',
|
|
147
|
+
},
|
|
148
|
+
], 4000)
|
|
149
|
+
|
|
150
|
+
await z.sleep(1000)
|
|
151
|
+
|
|
152
|
+
sip.call.stop_record_wav(oc.id)
|
|
153
|
+
sip.call.stop_record_wav(ic.id)
|
|
154
|
+
|
|
155
|
+
sip.call.terminate(oc.id)
|
|
156
|
+
|
|
157
|
+
await z.wait([
|
|
158
|
+
{
|
|
159
|
+
event: 'call_ended',
|
|
160
|
+
call_id: oc.id,
|
|
161
|
+
},
|
|
162
|
+
{
|
|
163
|
+
event: 'call_ended',
|
|
164
|
+
call_id: ic.id,
|
|
165
|
+
},
|
|
166
|
+
{
|
|
167
|
+
event: 'response',
|
|
168
|
+
call_id: oc.id,
|
|
169
|
+
method: 'BYE',
|
|
170
|
+
msg: sip_msg({
|
|
171
|
+
$rs: '200',
|
|
172
|
+
$rr: 'OK',
|
|
173
|
+
}),
|
|
174
|
+
},
|
|
175
|
+
], 1000)
|
|
176
|
+
|
|
177
|
+
await z.sleep(100)
|
|
178
|
+
|
|
179
|
+
console.log("Success")
|
|
180
|
+
|
|
181
|
+
sip.stop()
|
|
182
|
+
}
|
|
183
|
+
|
|
184
|
+
|
|
185
|
+
test()
|
|
186
|
+
.catch(e => {
|
|
187
|
+
console.error(e)
|
|
188
|
+
process.exit(1)
|
|
189
|
+
})
|
|
190
|
+
|