sip-lab 1.15.0 → 1.16.0

package/samples/rtp_and_srtp.rtp_refused.js

@@ -0,0 +1,430 @@
+const sip = require ('../index.js')
+const Zeq = require('@mayama/zeq')
+const m = require('data-matching')
+const sip_msg = require('sip-matching')
+const sdp = require('sdp-matching')
+
+// here we create our Zeq instance
+var z = new Zeq()
+
+async function test() {
+    sip.dtmf_aggregation_on(500)
+
+    z.trap_events(sip.event_source, 'event', (evt) => {
+        var e = evt.args[0]
+        return e
+    })
+
+    // Let's ignore '100 Trying'
+    z.add_event_filter({
+        event: 'response',
+        msg: sip_msg({
+            $rs: '100',
+        }),
+    })
+
+    console.log(sip.start((data) => { console.log(data)} ))
+
+    const t1 = sip.transport.create({address: "127.0.0.1"})
+    const t2 = sip.transport.create({address: "127.0.0.1"})
+
+    console.log("t1", t1)
+    console.log("t2", t2)
+
+    const oc = sip.call.create(t1.id, {from_uri: 'sip:alice@test.com', to_uri: `sip:bob@${t2.address}:${t2.port}`,
+        media: [
+            {
+                type: 'audio',
+                secure: false,
+            },
+            {
+                type: 'audio',
+                secure: true,
+            },
+        ]})
+
+    await z.wait([
+        {
+            event: "incoming_call",
+            call_id: m.collect("call_id"),
+            transport_id: t2.id,
+        },
+    ], 1000)
+
+    const ic = {
+        id: z.store.call_id,
+        sip_call_id: z.store.sip_call_id,
+    }
+
+    sip.call.respond(ic.id, {code: 200, reason: 'OK', media: [
+        {
+            type: 'audio',
+            port: 0, // media refused
+        },
+        {
+            type: 'audio',
+            secure: true,
+        },
+    ]})
+
+    await z.wait([
+        {
+            event: 'response',
+            call_id: oc.id,
+            method: 'INVITE',
+            msg: sip_msg({
+                $rs: '200',
+                $rr: 'OK',
+                '$(hdrcnt(VIA))': 1,
+                $fU: 'alice',
+                $fd: 'test.com',
+                $tU: 'bob',
+                '$hdr(content-type)': 'application/sdp',
+                $rb: '!{_}a=sendrecv',
+            }),
+        },
+        {
+            event: 'media_update',
+            call_id: oc.id,
+            status: 'ok',
+            media: m.fm([
+              m.pm({
+                type: 'audio',
+                protocol: 'RTP/AVP',
+                port: 0,
+              }),
+              m.pm({
+                type: 'audio',
+                protocol: 'RTP/SAVP',
+                local: {
+                  mode: 'sendrecv'
+                },
+                remote: {
+                  mode: 'sendrecv'
+                },
+                fmt: [
+                  '0 PCMU/8000',
+                  '120 telephone-event/8000'
+                ]
+              }),
+            ]),
+        },
+        {
+            event: 'media_update',
+            call_id: ic.id,
+            status: 'ok',
+            media: m.fm([
+              m.pm({
+                type: 'audio',
+                protocol: 'RTP/AVP',
+                port: 0,
+              }),
+              m.pm({
+                type: 'audio',
+                protocol: 'RTP/SAVP',
+                local: {
+                  mode: 'sendrecv'
+                },
+                remote: {
+                  mode: 'sendrecv'
+                },
+                fmt: [
+                  '0 PCMU/8000',
+                  '120 telephone-event/8000'
+                ]
+              }),
+            ]),
+        },
+    ], 1000)
+
+    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: 1,
+        },
+        {
+            event: 'dtmf',
+            call_id: oc.id,
+            digits: '1234',
+            mode: 1,
+            media_id: 1,
+        },
+    ], 2000)
+
+    sip.call.reinvite(oc.id, {media: [
+        {
+            type: 'audio',
+            secure: false,
+        },
+        {
+            type: 'audio',
+            secure: true,
+        },
+    ]})
+
+    await z.wait([
+        {
+            event: 'reinvite',
+            call_id: ic.id,
+        },
+    ], 500)
+
+    sip.call.respond(ic.id, {code: 200, reason: 'OK', media: [
+        {
+            type: 'audio',
+            port: 0, // media refused
+        },
+        {
+            type: 'audio',
+            secure: true,
+        },
+    ]})
+
+    await z.wait([
+        {
+            event: 'response',
+            call_id: oc.id,
+            method: 'INVITE',
+            msg: sip_msg({
+                $rs: '200',
+                $rb: sdp.jsonpath_matcher({
+                    '$.media.length': [2],
+                    '$.media[*].desc.type': ['audio', 'audio'],
+                    '$.media[*].desc.port': ['0', m.nonzero],
+                    '$.media[*].desc.protocol': ['RTP/AVP', 'RTP/SAVP'],
+                }),
+            }),
+        },
+        {
+            event: 'media_update',
+            call_id: ic.id,
+            status: 'ok',
+            media: m.fm([
+              m.pm({
+                type: 'audio',
+                protocol: 'RTP/AVP',
+                port: 0,
+              }),
+              m.pm({
+                type: 'audio',
+                protocol: 'RTP/SAVP',
+                local: {
+                  mode: 'sendrecv'
+                },
+                remote: {
+                  mode: 'sendrecv'
+                },
+                fmt: [
+                  '0 PCMU/8000',
+                  '120 telephone-event/8000'
+                ]
+              }),
+            ]),
+        },
+        {
+            event: 'media_update',
+            call_id: oc.id,
+            status: 'ok',
+            media: m.fm([
+              m.pm({
+                type: 'audio',
+                protocol: 'RTP/AVP',
+                port: 0,
+              }),
+              m.pm({
+                type: 'audio',
+                protocol: 'RTP/SAVP',
+                local: {
+                  mode: 'sendrecv'
+                },
+                remote: {
+                  mode: 'sendrecv'
+                },
+                fmt: [
+                  '0 PCMU/8000',
+                  '120 telephone-event/8000'
+                ]
+              }),
+            ]),
+        },
+    ], 1000)
+
+    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: 1,
+        },
+        {
+            event: 'dtmf',
+            call_id: oc.id,
+            digits: '1234',
+            mode: 1,
+            media_id: 1,
+        },
+    ], 2000)
+
+
+    sip.call.reinvite(ic.id, {media: [
+        {
+            type: 'audio',
+            secure: false,
+        },
+        {
+            type: 'audio',
+            secure: true,
+        },
+    ]})
+
+    await z.wait([
+        {
+            event: 'reinvite',
+            call_id: oc.id,
+        },
+    ], 500)
+
+    sip.call.respond(oc.id, {code: 200, reason: 'OK', media: [
+        {
+            type: 'audio',
+            port: 0, // media refused
+        },
+        {
+            type: 'audio',
+            secure: true,
+        },
+    ]})
+
+    await z.wait([
+        {
+            event: 'response',
+            call_id: ic.id,
+            method: 'INVITE',
+            msg: sip_msg({
+                $rs: '200',
+                $rb: sdp.jsonpath_matcher({
+                    '$.media.length': [2],
+                    '$.media[*].desc.type': ['audio', 'audio'],
+                    '$.media[*].desc.port': ['0', m.nonzero],
+                    '$.media[*].desc.protocol': ['RTP/AVP', 'RTP/SAVP'],
+                }),
+            }),
+        },
+        {
+            event: 'media_update',
+            call_id: oc.id,
+            status: 'ok',
+            media: m.fm([
+              m.pm({
+                type: 'audio',
+                protocol: 'RTP/AVP',
+                port: 0,
+              }),
+              m.pm({
+                type: 'audio',
+                protocol: 'RTP/SAVP',
+                local: {
+                  mode: 'sendrecv'
+                },
+                remote: {
+                  mode: 'sendrecv'
+                },
+                fmt: [
+                  '0 PCMU/8000',
+                  '120 telephone-event/8000'
+                ]
+              }),
+            ]),
+        },
+        {
+            event: 'media_update',
+            call_id: ic.id,
+            status: 'ok',
+            media: m.fm([
+              m.pm({
+                type: 'audio',
+                protocol: 'RTP/AVP',
+                port: 0,
+              }),
+              m.pm({
+                type: 'audio',
+                protocol: 'RTP/SAVP',
+                local: {
+                  mode: 'sendrecv'
+                },
+                remote: {
+                  mode: 'sendrecv'
+                },
+                fmt: [
+                  '0 PCMU/8000',
+                  '120 telephone-event/8000'
+                ]
+              }),
+            ]),
+        },
+    ], 1000)
+
+    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: 1,
+        },
+        {
+            event: 'dtmf',
+            call_id: oc.id,
+            digits: '1234',
+            mode: 1,
+            media_id: 1,
+        },
+    ], 2000)
+
+    sip.call.terminate(oc.id)
+
+    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,
+        },
+    ], 1000)
+
+    console.log("Success")
+
+    sip.stop()
+}
+
+
+test()
+.catch(e => {
+    console.error(e)
+    process.exit(1)
+})
+

package/samples/rtp_and_srtp.unbalanced_sdp_answer.js.future

@@ -0,0 +1,189 @@
+const sip = require ('../index.js')
+const Zeq = require('@mayama/zeq')
+const m = require('data-matching')
+const sip_msg = require('sip-matching')
+const sdp = require('sdp-matching')
+
+const sipjs = require('sipjs-lab')
+const {endpoint, dialog} = require('sipjs-lab')
+
+const util = require('util')
+
+// here we create our Zeq instance
+var z = new Zeq()
+
+async function test() {
+    sip.dtmf_aggregation_on(500)
+
+    z.trap_events(sip.event_source, 'event', (evt) => {
+        var e = evt.args[0]
+        return e
+    })
+
+    z.trap_events(sipjs.event_source, 'event', (evt) => {
+        var e = evt.args[0]
+        return e
+    })
+
+    // Let's ignore '100 Trying'
+    z.add_event_filter({
+        event: 'response',
+        msg: sip_msg({
+            $rs: '100',
+        }),
+    })
+
+    console.log(sip.start((data) => { console.log(data)} ))
+
+    const t1 = sip.transport.create({address: "127.0.0.1"})
+
+    const e1_port = 7070
+    const e1 = endpoint.create({address: "127.0.0.1", port: e1_port, publicAddress: "127.0.0.1" })
+
+    const oc = sip.call.create(t1.id, {from_uri: 'sip:alice@test.com', to_uri: `sip:bob@127.0.0.1:${e1_port}`,
+        media: [
+            {
+                type: 'audio',
+                secure: false,
+            },
+            {
+                type: 'audio',
+                secure: true,
+            },
+        ]})
+
+    await z.wait([
+        {
+            source: 'sip_endpoint',
+            endpoint_id: m.collect('endpoint_id'),
+            req: m.collect('req', {
+                method: 'INVITE',
+                uri: 'sip:bob@127.0.0.1:7070',
+                version: '2.0',
+                headers: {
+                    from: {
+                        uri: 'sip:alice@test.com',
+                    },
+                    to: {
+                        uri: 'sip:bob@127.0.0.1',
+                    },
+                },
+                //content: 
+            }),
+            event: 'dialog_offer',
+            dialog_id: m.collect('dialog_id'),
+        },
+    ], 1000)
+
+    const sdp_answer =`v=0
+o=- 3914707056 3914707057 IN IP4 0.0.0.0
+s=pjmedia
+t=0 0
+m=audio 10004 RTP/SAVP 0 120
+c=IN IP4 127.0.0.1
+b=TIAS:64000
+a=rtcp:10005 IN IP4 127.0.0.1
+a=sendrecv
+a=rtpmap:0 PCMU/8000
+a=rtpmap:120 telephone-event/8000
+a=fmtp:120 0-16
+a=crypto:1 AES_256_CM_HMAC_SHA1_80 inline:s6nwVuVNxwupG0Cls+/4X4L46Mx/fqoEMolllUrx/FNCFpVjG2kGFLxPRGvMNw==`
+
+    dialog.send_reply(
+        z.store.dialog_id,
+        z.store.req,
+        {
+            status: 200,
+            reason: 'OK',
+            headers: {
+                'content-type': 'application/sdp',
+            },
+            content: sdp_answer,
+        }
+    )
+
+    await z.wait([
+        {
+            event: 'response',
+            call_id: oc.id,
+            method: 'INVITE',
+            msg: sip_msg({
+                $rs: '200',
+                $rr: 'OK',
+                '$(hdrcnt(VIA))': 1,
+                $fU: 'alice',
+                $fd: 'test.com',
+                $tU: 'bob',
+                '$hdr(content-type)': 'application/sdp',
+                $rb: '!{_}a=sendrecv',
+            }),
+        },
+        {
+            event: 'media_update',
+            call_id: oc.id,
+            status: 'ok',
+            media: m.fm([
+              m.pm({
+                type: 'audio',
+                protocol: 'RTP/AVP',
+                port: 0,
+              }),
+              m.pm({
+                type: 'audio',
+                protocol: 'RTP/SAVP',
+                local: {
+                  mode: 'sendrecv'
+                },
+                remote: {
+                  mode: 'sendrecv'
+                },
+                fmt: [
+                  '0 PCMU/8000',
+                  '120 telephone-event/8000'
+                ]
+              }),
+            ]),
+        },
+        {
+            source: 'sip_endpoint',
+            endpoint_id: z.store.endpoint_id,
+            req: {
+              method: 'ACK',
+            },
+            event: 'in_dialog_request',
+            dialog_id: z.store.dialog_id
+        },
+    ], 1000)
+
+    await z.sleep(1000)
+
+    sip.call.terminate(oc.id)
+
+    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,
+        },
+    ], 1000)
+
+    console.log("Success")
+
+    sip.stop()
+}
+
+
+test()
+.catch(e => {
+    console.error(e)
+    process.exit(1)
+})
+

package/samples/srtp.js

@@ -1,5 +1,3 @@
-// This test creates 2 UDP SIP endpoints, makes a call between them and disconeects.
-
 const sip = require ('../index.js')
 const Zeq = require('@mayama/zeq')
 const m = require('data-matching')
@@ -55,9 +53,9 @@ async function test() {
     }
 
     sip.call.respond(ic.id, {code: 200, reason: 'OK', media: [
-	{
+	    {
             type: 'audio',
-	    secure: true,
+            secure: true,
         },
     ]})
 
@@ -110,9 +108,9 @@ async function test() {
     ], 2000)
 
     sip.call.reinvite(oc.id, {media: [
-	{
+        {
             type: 'audio',
-	    secure: true,
+            secure: true,
         },
     ]})
 
@@ -124,9 +122,9 @@ async function test() {
     ], 500)
 
     sip.call.respond(ic.id, {code: 200, reason: 'OK', media: [
-	{
+        {
             type: 'audio',
-	    secure: true,
+            secure: true,
         },
     ]})
 
@@ -152,7 +150,7 @@ async function test() {
             media: m.fm([
               m.pm({
                 type: 'audio',
-		transport: 'RTP/SAVP',
+                protocol: 'RTP/SAVP',
                 local: {
                   mode: 'sendrecv'
                 },
@@ -173,7 +171,7 @@ async function test() {
             media: m.fm([
               m.pm({
                 type: 'audio',
-		transport: 'RTP/SAVP',
+                protocol: 'RTP/SAVP',
                 local: {
                   mode: 'sendrecv'
                 },
@@ -211,9 +209,9 @@ async function test() {
 
 
     sip.call.reinvite(ic.id, {media: [
-	{
+        {
             type: 'audio',
-	    secure: true,
+            secure: true,
         },
     ]})
 
@@ -225,9 +223,9 @@ async function test() {
     ], 500)
 
     sip.call.respond(oc.id, {code: 200, reason: 'OK', media: [
-	{
+        {
             type: 'audio',
-	    secure: true,
+            secure: true,
         },
     ]})
 
@@ -253,7 +251,7 @@ async function test() {
             media: m.fm([
               m.pm({
                 type: 'audio',
-		transport: 'RTP/SAVP',
+                protocol: 'RTP/SAVP',
                 local: {
                   mode: 'sendrecv'
                 },
@@ -274,7 +272,7 @@ async function test() {
             media: m.fm([
               m.pm({
                 type: 'audio',
-		transport: 'RTP/SAVP',
+                protocol: 'RTP/SAVP',
                 local: {
                   mode: 'sendrecv'
                 },