sip-lab 1.15.0 → 1.17.0

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Files changed (9) hide show
  1. package/package.json +4 -3
  2. package/prebuilds/linux-x64/sip-lab.node +0 -0
  3. package/samples/16_calls.js +152 -0
  4. package/samples/{183_session_progress.js.future → 183_session_progress.js} +156 -7
  5. package/samples/rtp_and_srtp.js +466 -0
  6. package/samples/rtp_and_srtp.rtp_refused.js +430 -0
  7. package/samples/rtp_and_srtp.unbalanced_sdp_answer.js.future +189 -0
  8. package/samples/srtp.js +14 -16
  9. package/src/sip.cpp +53 -22
package/samples/rtp_and_srtp.js ADDED
@@ -0,0 +1,466 @@
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+ const sip = require ('../index.js')
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+ const Zeq = require('@mayama/zeq')
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+ const m = require('data-matching')
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+ const sip_msg = require('sip-matching')
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+ const sdp = require('sdp-matching')
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+
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+ // here we create our Zeq instance
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+ var z = new Zeq()
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+
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+ async function test() {
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+ sip.dtmf_aggregation_on(500)
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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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+
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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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+ $rs: '100',
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+ }),
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+ })
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+
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+ console.log(sip.start((data) => { console.log(data)} ))
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+
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+ const t1 = sip.transport.create({address: "127.0.0.1"})
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+ const t2 = sip.transport.create({address: "127.0.0.1"})
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+
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+ console.log("t1", t1)
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+ console.log("t2", t2)
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+
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+ const oc = sip.call.create(t1.id, {from_uri: 'sip:alice@test.com', to_uri: `sip:bob@${t2.address}:${t2.port}`,
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+ media: [
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+ {
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+ type: 'audio',
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+ secure: false,
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+ },
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+ {
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+ type: 'audio',
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+ secure: true,
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+ },
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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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+ transport_id: t2.id,
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+ },
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+ ], 1000)
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+
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+ const 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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+
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+ sip.call.respond(ic.id, {code: 200, reason: 'OK', media: [
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+ {
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+ type: 'audio',
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+ secure: false,
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+ },
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+ {
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+ type: 'audio',
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+ secure: true,
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+ },
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+ ]})
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+
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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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+ },
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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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+ ], 1000)
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+
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+ sip.call.send_dtmf(oc.id, {digits: '1234', mode: 1})
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+ sip.call.send_dtmf(ic.id, {digits: '1234', mode: 1})
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+
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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: 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: 1,
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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: '1234',
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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: oc.id,
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+ digits: '1234',
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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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+
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+ sip.call.reinvite(oc.id, {media: [
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+ {
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+ type: 'audio',
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+ secure: false,
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+ },
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+ {
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+ type: 'audio',
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+ secure: true,
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+ },
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+ ]})
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+
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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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+ ], 500)
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+
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+ sip.call.respond(ic.id, {code: 200, reason: 'OK', media: [
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+ {
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+ type: 'audio',
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+ secure: false,
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+ },
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+ {
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+ type: 'audio',
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+ secure: true,
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+ },
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+ ]})
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+
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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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+ $rb: sdp.jsonpath_matcher({
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+ '$.media.length': [2],
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+ '$.media[*].desc.type': ['audio', 'audio'],
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+ '$.media[*].desc.port': [m.nonzero, m.nonzero],
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+ '$.media[*].desc.protocol': ['RTP/AVP', 'RTP/SAVP'],
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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: m.fm([
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+ m.pm({
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+ type: 'audio',
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+ protocol: 'RTP/AVP',
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+ local: {
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+ mode: 'sendrecv'
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+ },
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+ remote: {
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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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+ m.pm({
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+ type: 'audio',
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+ protocol: 'RTP/SAVP',
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+ local: {
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+ mode: 'sendrecv'
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+ },
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+ remote: {
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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: m.fm([
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+ m.pm({
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+ type: 'audio',
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+ protocol: 'RTP/AVP',
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+ local: {
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+ mode: 'sendrecv'
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+ },
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+ remote: {
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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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+ m.pm({
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+ type: 'audio',
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+ protocol: 'RTP/SAVP',
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+ local: {
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+ mode: 'sendrecv'
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+ },
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+ remote: {
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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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+
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+ sip.call.send_dtmf(oc.id, {digits: '1234', mode: 1})
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+ sip.call.send_dtmf(ic.id, {digits: '1234', mode: 1})
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+
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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: 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: 1,
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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: '1234',
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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: oc.id,
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+ digits: '1234',
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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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+
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+
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+ sip.call.reinvite(ic.id, {media: [
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+ {
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+ type: 'audio',
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+ secure: false,
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+ },
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+ {
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+ type: 'audio',
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+ secure: true,
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+ },
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+ ]})
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+
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+ await z.wait([
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+ {
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+ event: 'reinvite',
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+ call_id: oc.id,
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+ },
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+ ], 500)
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+
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+ sip.call.respond(oc.id, {code: 200, reason: 'OK', media: [
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+ {
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+ type: 'audio',
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+ secure: false,
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+ },
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+ {
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+ type: 'audio',
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+ secure: true,
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+ },
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+ ]})
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+
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+ await z.wait([
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+ {
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+ event: 'response',
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+ call_id: ic.id,
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+ method: 'INVITE',
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+ msg: sip_msg({
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+ $rs: '200',
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+ $rb: sdp.jsonpath_matcher({
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+ '$.media.length': [2],
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+ '$.media[*].desc.type': ['audio', 'audio'],
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+ '$.media[*].desc.port': [m.nonzero, m.nonzero],
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+ '$.media[*].desc.protocol': ['RTP/AVP', 'RTP/SAVP'],
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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: m.fm([
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+ m.pm({
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+ type: 'audio',
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+ protocol: 'RTP/AVP',
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+ local: {
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+ mode: 'sendrecv'
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+ },
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+ remote: {
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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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+ m.pm({
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+ type: 'audio',
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+ protocol: 'RTP/SAVP',
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+ local: {
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+ mode: 'sendrecv'
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+ },
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+ remote: {
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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: ic.id,
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+ status: 'ok',
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+ media: m.fm([
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+ m.pm({
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+ type: 'audio',
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+ protocol: 'RTP/AVP',
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+ local: {
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+ mode: 'sendrecv'
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+ },
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+ remote: {
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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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+ m.pm({
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+ type: 'audio',
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+ protocol: 'RTP/SAVP',
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+ local: {
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+ mode: 'sendrecv'
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+ },
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+ remote: {
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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)
398
+
399
+ sip.call.send_dtmf(oc.id, {digits: '1234', mode: 1})
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+ sip.call.send_dtmf(ic.id, {digits: '1234', mode: 1})
401
+
402
+ await z.wait([
403
+ {
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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: 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: 1,
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+ media_id: 1,
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+ },
417
+ {
418
+ event: 'dtmf',
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+ call_id: oc.id,
420
+ digits: '1234',
421
+ mode: 1,
422
+ media_id: 0,
423
+ },
424
+ {
425
+ event: 'dtmf',
426
+ call_id: oc.id,
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+ digits: '1234',
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+ mode: 1,
429
+ media_id: 1,
430
+ },
431
+ ], 2000)
432
+
433
+ sip.call.terminate(oc.id)
434
+
435
+ await z.wait([
436
+ {
437
+ event: 'response',
438
+ call_id: oc.id,
439
+ method: 'BYE',
440
+ msg: sip_msg({
441
+ $rs: '200',
442
+ $rr: 'OK',
443
+ }),
444
+ },
445
+ {
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+ event: 'call_ended',
447
+ call_id: oc.id,
448
+ },
449
+ {
450
+ event: 'call_ended',
451
+ call_id: ic.id,
452
+ },
453
+ ], 1000)
454
+
455
+ console.log("Success")
456
+
457
+ sip.stop()
458
+ }
459
+
460
+
461
+ test()
462
+ .catch(e => {
463
+ console.error(e)
464
+ process.exit(1)
465
+ })
466
+