sip-lab 1.12.39 → 1.13.1
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/DEV.md +19 -3
- package/README.md +8 -4
- package/binding.gyp +97 -86
- package/build_deps.sh +3 -2
- package/index.js +14 -7
- package/package.json +7 -2
- package/prebuilds/linux-x64/node.abi102.node +0 -0
- package/prebuilds/linux-x64/node.abi108.node +0 -0
- package/prebuilds/linux-x64/node.abi88.node +0 -0
- package/prebuilds/linux-x64/node.abi93.node +0 -0
- package/samples/16_audio_streams.js +179 -0
- package/samples/delayed_media.js +117 -81
- package/samples/g729.js +50 -28
- package/samples/mrcp_and_audio.js +445 -0
- package/samples/mrcp_and_audio.simplified_media.js +273 -0
- package/samples/mrcp_and_audio.switching_order.js +273 -0
- package/samples/options.js +82 -0
- package/samples/refer.js +247 -0
- package/samples/refuse_telephone_event.js +166 -0
- package/samples/register_no_expires.js +82 -0
- package/samples/register_subscribe.js +36 -4
- package/samples/reinvite_and_dtmf.js +236 -161
- package/samples/reinvite_audio_audio.js +320 -0
- package/samples/reinvite_with_hold_unhold.js +412 -0
- package/samples/send_and_receive_fax.js +4 -8
- package/samples/simple.js +5 -9
- package/samples/sip_cancel.js +4 -6
- package/samples/tcp_and_extra_headers.js +98 -45
- package/samples/two_audio_media.js +497 -0
- package/src/addon.cpp +488 -268
- package/src/event_templates.cpp +84 -35
- package/src/event_templates.hpp +22 -12
- package/src/idmanager.cpp +58 -57
- package/src/idmanager.hpp +10 -10
- package/src/log.cpp +9 -11
- package/src/log.hpp +4 -4
- package/src/sip.cpp +6841 -5040
- package/src/sip.hpp +26 -34
- package/src/packetdumper.cpp +0 -234
- package/src/packetdumper.hpp +0 -67
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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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async function test() {
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sip.set_log_level(9)
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//sip.set_log_level(6)
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sip.dtmf_aggregation_on(500)
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// Let's ignore '100 Trying'
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z.add_event_filter({
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event: 'response',
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msg: sip_msg({
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$rs: '100',
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}),
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})
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z.trap_events(sip.event_source, 'event', (evt) => {
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var e = evt.args[0]
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return e
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})
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console.log(sip.start((data) => { console.log(data)} ))
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t1 = sip.transport.create({address: "127.0.0.1", port: 5090, type: 'udp'})
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t2 = sip.transport.create({address: "127.0.0.1", port: 5092, type: 'udp'})
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console.log("t1", t1)
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console.log("t2", t2)
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oc = sip.call.create(t1.id, {from_uri: 'sip:alice@test.com', to_uri: `sip:bob@${t2.address}:${t2.port}`, media: "audio,audio"})
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await z.wait([
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{
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event: "incoming_call",
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call_id: m.collect("call_id"),
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},
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], 1000)
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ic = {
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id: z.store.call_id,
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sip_call_id: z.store.sip_call_id,
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}
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sip.call.respond(ic.id, {code: 200, reason: 'OK', media: "audio,audio"})
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await z.wait([
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{
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event: 'response',
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call_id: oc.id,
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method: 'INVITE',
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msg: sip_msg({
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$rs: '200',
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$rr: 'OK',
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'$(hdrcnt(VIA))': 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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}),
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},
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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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media: [
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{
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type: 'audio',
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},
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{
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type: 'audio',
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}
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],
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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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media: [
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{
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type: 'audio',
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},
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{
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type: 'audio',
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}
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],
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},
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], 1000)
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sip.call.send_dtmf(oc.id, {digits: '1234', mode: 0})
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sip.call.send_dtmf(ic.id, {digits: '4321', mode: 1})
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await z.wait([
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{
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event: 'dtmf',
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call_id: ic.id,
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digits: '1234',
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mode: 0,
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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: oc.id,
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digits: '4321',
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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: '1234',
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mode: 0,
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media_id: 1
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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: '4321',
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mode: 1,
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media_id: 1
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},
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], 2000)
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sip.call.reinvite(oc.id, {media: "audio,audio"})
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await z.wait([
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{
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event: 'reinvite',
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call_id: ic.id,
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},
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], 1000)
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sip.call.respond(ic.id, {code: 200, reason: 'OK', media: "audio,audio"})
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await z.wait([
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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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}),
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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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media: [
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{
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type: 'audio',
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local: {
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addr: '127.0.0.1',
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port: 10004,
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mode: 'sendrecv'
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},
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remote: {
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addr: '127.0.0.1',
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port: 10000,
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mode: 'sendrecv'
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},
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fmt: [
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'0 PCMU/8000',
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'120 telephone-event/8000'
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]
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},
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{
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type: 'audio',
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local: {
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addr: '127.0.0.1',
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port: 10006,
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mode: 'sendrecv'
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},
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remote: {
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addr: '127.0.0.1',
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port: 10002,
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mode: 'sendrecv'
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},
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fmt: [
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'0 PCMU/8000',
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'120 telephone-event/8000'
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]
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}
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]
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},
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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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media: [
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{
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type: 'audio',
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local: {
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addr: '127.0.0.1',
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port: 10000,
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mode: 'sendrecv'
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},
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remote: {
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addr: '127.0.0.1',
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port: 10004,
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mode: 'sendrecv'
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},
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fmt: [
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'0 PCMU/8000',
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'120 telephone-event/8000'
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]
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},
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{
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type: 'audio',
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local: {
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addr: '127.0.0.1',
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port: 10002,
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mode: 'sendrecv'
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},
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remote: {
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addr: '127.0.0.1',
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port: 10006,
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mode: 'sendrecv'
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},
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fmt: [
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'0 PCMU/8000',
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'120 telephone-event/8000'
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]
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}
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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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// Now change to single media
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sip.call.reinvite(oc.id, {media: "audio"})
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await z.wait([
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{
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event: 'reinvite',
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call_id: ic.id,
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},
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], 1000)
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sip.call.respond(ic.id, {code: 200, reason: 'OK', media: "audio,audio"})
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await z.wait([
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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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}),
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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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media: [
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{
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type: 'audio',
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local: {
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addr: '127.0.0.1',
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port: 10004,
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mode: 'sendrecv'
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},
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remote: {
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addr: '127.0.0.1',
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port: 10000,
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mode: 'sendrecv'
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},
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fmt: [
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'0 PCMU/8000',
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'120 telephone-event/8000'
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]
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},
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]
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},
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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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media: [
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{
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type: 'audio',
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local: {
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addr: '127.0.0.1',
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port: 10000,
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mode: 'sendrecv'
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},
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remote: {
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addr: '127.0.0.1',
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port: 10004,
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mode: 'sendrecv'
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},
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fmt: [
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'0 PCMU/8000',
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'120 telephone-event/8000'
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]
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},
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]
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},
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], 1000)
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sip.call.send_dtmf(oc.id, {digits: '1234', mode: 0})
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sip.call.send_dtmf(ic.id, {digits: '4321', mode: 1})
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await z.wait([
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{
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event: 'dtmf',
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call_id: ic.id,
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digits: '1234',
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mode: 0,
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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: oc.id,
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digits: '4321',
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mode: 1,
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media_id: 0
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},
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], 1500)
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// now switch back to two media
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sip.call.reinvite(oc.id, {media: "audio,audio"})
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328
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await z.wait([
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{
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event: 'reinvite',
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call_id: ic.id,
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},
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], 1000)
|
|
334
|
+
|
|
335
|
+
sip.call.respond(ic.id, {code: 200, reason: 'OK', media: "audio,audio"})
|
|
336
|
+
|
|
337
|
+
await z.wait([
|
|
338
|
+
{
|
|
339
|
+
event: 'response',
|
|
340
|
+
call_id: oc.id,
|
|
341
|
+
method: 'INVITE',
|
|
342
|
+
msg: sip_msg({
|
|
343
|
+
$rs: '200',
|
|
344
|
+
}),
|
|
345
|
+
},
|
|
346
|
+
{
|
|
347
|
+
event: 'media_update',
|
|
348
|
+
call_id: ic.id,
|
|
349
|
+
status: 'ok',
|
|
350
|
+
media: [
|
|
351
|
+
{
|
|
352
|
+
type: 'audio',
|
|
353
|
+
local: {
|
|
354
|
+
addr: '127.0.0.1',
|
|
355
|
+
port: 10004,
|
|
356
|
+
mode: 'sendrecv'
|
|
357
|
+
},
|
|
358
|
+
remote: {
|
|
359
|
+
addr: '127.0.0.1',
|
|
360
|
+
port: 10000,
|
|
361
|
+
mode: 'sendrecv'
|
|
362
|
+
},
|
|
363
|
+
fmt: [
|
|
364
|
+
'0 PCMU/8000',
|
|
365
|
+
'120 telephone-event/8000'
|
|
366
|
+
]
|
|
367
|
+
},
|
|
368
|
+
{
|
|
369
|
+
type: 'audio',
|
|
370
|
+
local: {
|
|
371
|
+
addr: '127.0.0.1',
|
|
372
|
+
port: 10006,
|
|
373
|
+
mode: 'sendrecv'
|
|
374
|
+
},
|
|
375
|
+
remote: {
|
|
376
|
+
addr: '127.0.0.1',
|
|
377
|
+
port: 10002,
|
|
378
|
+
mode: 'sendrecv'
|
|
379
|
+
},
|
|
380
|
+
fmt: [
|
|
381
|
+
'0 PCMU/8000',
|
|
382
|
+
'120 telephone-event/8000'
|
|
383
|
+
]
|
|
384
|
+
}
|
|
385
|
+
]
|
|
386
|
+
},
|
|
387
|
+
{
|
|
388
|
+
event: 'media_update',
|
|
389
|
+
call_id: oc.id,
|
|
390
|
+
status: 'ok',
|
|
391
|
+
media: [
|
|
392
|
+
{
|
|
393
|
+
type: 'audio',
|
|
394
|
+
local: {
|
|
395
|
+
addr: '127.0.0.1',
|
|
396
|
+
port: 10000,
|
|
397
|
+
mode: 'sendrecv'
|
|
398
|
+
},
|
|
399
|
+
remote: {
|
|
400
|
+
addr: '127.0.0.1',
|
|
401
|
+
port: 10004,
|
|
402
|
+
mode: 'sendrecv'
|
|
403
|
+
},
|
|
404
|
+
fmt: [
|
|
405
|
+
'0 PCMU/8000',
|
|
406
|
+
'120 telephone-event/8000'
|
|
407
|
+
]
|
|
408
|
+
},
|
|
409
|
+
{
|
|
410
|
+
type: 'audio',
|
|
411
|
+
local: {
|
|
412
|
+
addr: '127.0.0.1',
|
|
413
|
+
port: 10002,
|
|
414
|
+
mode: 'sendrecv'
|
|
415
|
+
},
|
|
416
|
+
remote: {
|
|
417
|
+
addr: '127.0.0.1',
|
|
418
|
+
port: 10006,
|
|
419
|
+
mode: 'sendrecv'
|
|
420
|
+
},
|
|
421
|
+
fmt: [
|
|
422
|
+
'0 PCMU/8000',
|
|
423
|
+
'120 telephone-event/8000'
|
|
424
|
+
]
|
|
425
|
+
}
|
|
426
|
+
]
|
|
427
|
+
},
|
|
428
|
+
], 1000)
|
|
429
|
+
|
|
430
|
+
sip.call.send_dtmf(oc.id, {digits: '1234', mode: 0})
|
|
431
|
+
sip.call.send_dtmf(ic.id, {digits: '4321', mode: 1})
|
|
432
|
+
|
|
433
|
+
await z.wait([
|
|
434
|
+
{
|
|
435
|
+
event: 'dtmf',
|
|
436
|
+
call_id: ic.id,
|
|
437
|
+
digits: '1234',
|
|
438
|
+
mode: 0,
|
|
439
|
+
media_id: 0
|
|
440
|
+
},
|
|
441
|
+
{
|
|
442
|
+
event: 'dtmf',
|
|
443
|
+
call_id: oc.id,
|
|
444
|
+
digits: '4321',
|
|
445
|
+
mode: 1,
|
|
446
|
+
media_id: 0
|
|
447
|
+
},
|
|
448
|
+
{
|
|
449
|
+
event: 'dtmf',
|
|
450
|
+
call_id: ic.id,
|
|
451
|
+
digits: '1234',
|
|
452
|
+
mode: 0,
|
|
453
|
+
media_id: 1
|
|
454
|
+
},
|
|
455
|
+
{
|
|
456
|
+
event: 'dtmf',
|
|
457
|
+
call_id: oc.id,
|
|
458
|
+
digits: '4321',
|
|
459
|
+
mode: 1,
|
|
460
|
+
media_id: 1
|
|
461
|
+
},
|
|
462
|
+
], 2000)
|
|
463
|
+
|
|
464
|
+
sip.call.terminate(oc.id)
|
|
465
|
+
|
|
466
|
+
await z.wait([
|
|
467
|
+
{
|
|
468
|
+
event: 'call_ended',
|
|
469
|
+
call_id: oc.id,
|
|
470
|
+
},
|
|
471
|
+
{
|
|
472
|
+
event: 'call_ended',
|
|
473
|
+
call_id: ic.id,
|
|
474
|
+
},
|
|
475
|
+
{
|
|
476
|
+
event: 'response',
|
|
477
|
+
call_id: oc.id,
|
|
478
|
+
method: 'BYE',
|
|
479
|
+
msg: sip_msg({
|
|
480
|
+
$rs: '200',
|
|
481
|
+
$rr: 'OK',
|
|
482
|
+
}),
|
|
483
|
+
},
|
|
484
|
+
], 1000)
|
|
485
|
+
|
|
486
|
+
console.log("Success")
|
|
487
|
+
|
|
488
|
+
sip.stop()
|
|
489
|
+
}
|
|
490
|
+
|
|
491
|
+
|
|
492
|
+
test()
|
|
493
|
+
.catch(e => {
|
|
494
|
+
console.error(e)
|
|
495
|
+
process.exit(1)
|
|
496
|
+
})
|
|
497
|
+
|