sip-lab 1.38.0 → 1.40.0

package/samples/t

@@ -0,0 +1,239 @@
+var sip = require ('../index.js')
+var Zeq = require('@mayama/zeq')
+var z = new Zeq()
+var m = require('data-matching')
+var sip_msg = require('sip-matching')
+var sdp = require('sdp-matching')
+
+var assert = require('assert')
+
+async function test() {
+    //sip.set_log_level(6)
+    sip.dtmf_aggregation_on(500)
+
+    z.trap_events(sip.event_source, 'event', (evt) => {
+        var e = evt.args[0]
+        return e
+    })
+
+    console.log(sip.start((data) => { console.log(data)} ))
+
+    t1 = sip.transport.create({address: "127.0.0.1", type: 'udp'})
+    t2 = sip.transport.create({address: "127.0.0.1", type: 'udp'})
+
+    console.log("t1", t1)
+    console.log("t2", t2)
+
+    console.log(sip.get_codecs())
+    sip.set_codecs("opus/48000/2:128")
+
+    flags = 0
+
+    oc = sip.call.create(t1.id, {from_uri: 'sip:alice@test.com', to_uri: `sip:bob@${t2.address}:${t2.port}`})
+
+    await z.wait([
+        {
+            event: "incoming_call",
+            call_id: m.collect("call_id"),
+        },
+        {
+            event: 'response',
+            call_id: oc.id,
+            method: 'INVITE',
+            msg: sip_msg({
+                $rs: '100',
+                $rr: 'Trying',
+            }),
+        },
+    ], 1000)
+
+    ic = {
+        id: z.$call_id,
+        sip_call_id: z.$sip_call_id,
+    }
+
+    sip.call.respond(ic.id, {code: 200, reason: 'OK'})
+
+    await z.wait([
+        {
+            event: 'media_update',
+            call_id: oc.id,
+            status: 'ok',
+        },
+        {
+            event: 'media_update',
+            call_id: ic.id,
+            status: 'ok',
+        },
+        {
+            event: 'response',
+            call_id: oc.id,
+            method: 'INVITE',
+            msg: sip_msg({
+                $rs: '200',
+                $rr: 'OK',
+                'hdr_content_type': 'application/sdp',
+                $rb: '!{_}a=sendrecv!{_}',
+            }),
+        },
+    ], 1000)
+
+    sip.call.start_record_wav(oc.id, {file: './oc.wav'})
+    sip.call.start_record_wav(ic.id, {file: './ic.wav'})
+
+    sip.call.start_play_wav(oc.id, {file: 'samples/artifacts/hello_good_morning.wav', end_of_file_event: true, no_loop: true})
+    sip.call.start_play_wav(ic.id, {file: 'samples/artifacts/hello_good_morning.wav', end_of_file_event: true, no_loop: true})
+
+    await z.wait([
+        {
+            event: 'end_of_file',
+            call_id: oc.id,
+        },
+        {
+            event: 'end_of_file',
+            call_id: ic.id,
+        },
+    ], 5000)
+
+    sip.call.reinvite(oc.id)
+
+    await z.wait([
+        {
+            event: 'reinvite',
+            call_id: ic.id
+        },
+    ], 1000)
+
+    sip.call.respond(ic.id, {code: 200, reason: 'OK'})
+
+    await z.wait([
+        {
+            event: 'response',
+            call_id: oc.id,
+            method: 'INVITE',
+            msg: sip_msg({
+                $rs: '100',
+            }),
+        },
+        {
+            event: 'response',
+            call_id: oc.id,
+            method: 'INVITE',
+            msg: sip_msg({
+                $rs: '200',
+                $rr: 'OK',
+                $rb: '!{_}a=sendrecv!{_}',
+            }),
+        },
+        {
+            event: 'media_update',
+            call_id: oc.id,
+            status: 'ok',
+        },
+        {
+            event: 'media_update',
+            call_id: ic.id,
+            status: 'ok',
+        },
+    ], 500)
+
+    sip.call.reinvite(oc.id, false, 0)
+
+    await z.wait([
+        {
+            event: 'reinvite',
+            call_id: ic.id
+        },
+    ], 1000)
+
+    sip.call.respond(ic.id, {code: 200, reason: 'OK'})
+
+    await z.wait([
+        {
+            event: 'response',
+            call_id: oc.id,
+            method: 'INVITE',
+            msg: sip_msg({
+                $rs: '100',
+            }),
+        },
+        {
+            event: 'response',
+            call_id: oc.id,
+            method: 'INVITE',
+            msg: sip_msg({
+                $rs: '200',
+                $rr: 'OK',
+                $rb: '!{_}a=sendrecv!{_}',
+            }),
+        },
+        {
+            event: 'media_update',
+            call_id: oc.id,
+            status: 'ok',
+        },
+        {
+            event: 'media_update',
+            call_id: ic.id,
+            status: 'ok',
+        },
+    ], 500)
+
+    oc_stat = sip.call.get_stream_stat(oc.id, {media_id: 0})
+    ic_stat = sip.call.get_stream_stat(ic.id, {media_id: 0})
+
+    console.log(oc_stat)
+    console.log(ic_stat)
+
+    oc_stat = JSON.parse(oc_stat)
+    ic_stat = JSON.parse(ic_stat)
+
+    assert(oc_stat.CodecInfo == 'opus/8000/1')
+    assert(ic_stat.CodecInfo == 'opus/8000/1')
+
+    await z.sleep(100)
+
+    sip.call.start_inband_dtmf_detection(oc.id)
+    sip.call.start_inband_dtmf_detection(ic.id)
+
+    sip.call.send_dtmf(oc.id, {digits: '12', mode: 1})
+    sip.call.send_dtmf(ic.id, {digits: '21', mode: 1})
+
+    await z.sleep(1000)
+
+    sip.call.stop_record_wav(oc.id)
+    sip.call.stop_record_wav(ic.id)
+
+    sip.call.terminate(oc.id)
+
+    await z.wait([
+        {
+            event: 'call_ended',
+            call_id: oc.id,
+        },
+        {
+            event: 'call_ended',
+            call_id: ic.id,
+        },
+        {
+            event: 'response',
+            call_id: oc.id,
+            method: 'BYE',
+            msg: sip_msg({
+                $rs: '200',
+                $rr: 'OK',
+            }),
+        },
+    ], 1000)
+
+    console.log("Success")
+
+    sip.stop()
+    process.exit(0)
+}
+
+test()
+.catch(e => {
+    console.error(e)
+    process.exit(1)
+})

package/samples/websocket.js

@@ -0,0 +1,156 @@
+const sip = require('../index.js')
+const Zeq = require('@mayama/zeq')
+const m = require('data-matching')
+const sip_msg = require('sip-matching')
+
+var z = new Zeq()
+
+async function test() {
+    sip.dtmf_aggregation_on(500)
+
+    z.trap_events(sip.event_source, 'event', (evt) => {
+        return evt.args[0]
+    })
+
+    sip.set_codecs("pcmu/8000/1:128,pcma/8000/1:128")
+
+    console.log(sip.start((data) => { console.log(data) }))
+
+    // Create a WebSocket server transport (listener)
+    const t2 = sip.transport.create({
+        address: "127.0.0.1",
+        port: 6666,
+        type: "ws"
+    })
+
+    // Create a WebSocket client transport connecting to our server
+    const t1 = sip.transport.create({
+        address: "127.0.0.1",
+        type: "ws",
+        ws_url: "ws://127.0.0.1:6666/sip"
+    })
+
+    console.log("t1", t1)
+    console.log("t2", t2)
+
+    // Make the call from t1 to t2 over WebSocket
+    const oc = sip.call.create(t1.id, {
+        from_uri: 'sip:alice@test.com',
+        to_uri: 'sip:bob@127.0.0.1:8080',
+    })
+
+    // Wait for the call to arrive at t2 and 100 Trying response at t1
+    await z.wait([
+        {
+            event: "incoming_call",
+            call_id: m.collect("call_id"),
+            transport_id: t2.id,
+            msg: sip_msg({
+                $rU: 'bob',
+                $fU: 'alice',
+                $tU: 'bob',
+                $fd: 'test.com',
+            })
+        },
+        {
+            event: 'response',
+            call_id: oc.id,
+            method: 'INVITE',
+            msg: sip_msg({
+                $rs: '100',
+                $rr: 'Trying',
+            }),
+        },
+    ], 2000)
+
+    const ic = {
+        id: z.$call_id,
+        sip_call_id: z.$sip_call_id,
+    }
+
+    // Answer the call at t2 side
+    sip.call.respond(ic.id, {
+        code: 200,
+        reason: 'OK',
+    })
+
+    // Wait for 200 OK at t1 side and media setups
+    await z.wait([
+        {
+            event: 'response',
+            call_id: oc.id,
+            method: 'INVITE',
+            msg: sip_msg({
+                $rs: '200',
+                $rr: 'OK',
+            }),
+        },
+        {
+            event: 'media_update',
+            call_id: oc.id,
+            status: 'ok',
+        },
+        {
+            event: 'media_update',
+            call_id: ic.id,
+            status: 'ok',
+        },
+    ], 2000)
+
+    sip.call.start_inband_dtmf_detection(oc.id)
+    sip.call.start_inband_dtmf_detection(ic.id)
+
+    sip.call.send_dtmf(oc.id, {digits: '1234', mode: 1})
+    sip.call.send_dtmf(ic.id, {digits: '1234', mode: 1})
+
+    await z.wait([
+        {
+            event: 'dtmf',
+            call_id: ic.id,
+            digits: '1234',
+            mode: 1,
+            media_id: 0,
+        },
+        {
+            event: 'dtmf',
+            call_id: oc.id,
+            digits: '1234',
+            mode: 1,
+            media_id: 0,
+        },
+    ], 2000)
+
+    // Terminate the call from t1 side
+    sip.call.terminate(oc.id)
+
+    // Wait for call termination
+    await z.wait([
+        {
+            event: 'response',
+            call_id: oc.id,
+            method: 'BYE',
+            msg: sip_msg({
+                $rs: '200',
+                $rr: 'OK',
+            }),
+        },
+        {
+            event: 'call_ended',
+            call_id: oc.id,
+        },
+        {
+            event: 'call_ended',
+            call_id: ic.id,
+        },
+    ], 2000)
+
+    console.log("WebSocket test successful")
+
+    sip.stop()
+    process.exit(0)
+}
+
+test().catch(e => {
+    console.error(e)
+    process.exit(1)
+})

package/src/addon.cpp

@@ -1264,6 +1264,32 @@ Napi::Value set_codecs(const Napi::CallbackInfo &info) {
   return env.Null();
 }
 
+Napi::Value set_opus_config(const Napi::CallbackInfo &info) {
+  Napi::Env env = info.Env();
+
+  if (info.Length() != 1) {
+    Napi::Error::New(env, "Wrong number of arguments. Expected: config")
+        .ThrowAsJavaScriptException();
+    return env.Null();
+  }
+
+  if (!info[0].IsString()) {
+    Napi::TypeError::New(env, "config must be a JSON string.")
+        .ThrowAsJavaScriptException();
+    return env.Null();
+  }
+  string json = info[0].As<Napi::String>().Utf8Value();
+
+  int res = pjw_set_opus_config(json.c_str());
+
+  if (res != 0) {
+    Napi::Error::New(env, pjw_get_error()).ThrowAsJavaScriptException();
+    return env.Null();
+  }
+
+  return env.Null();
+}
+
 Napi::Value notify(const Napi::CallbackInfo &info) {
   Napi::Env env = info.Env();
 
@@ -1649,6 +1675,7 @@ Napi::Object init(Napi::Env env, Napi::Object exports) {
 
   exports.Set("get_codecs", Napi::Function::New(env, get_codecs));
   exports.Set("set_codecs", Napi::Function::New(env, set_codecs));
+  exports.Set("_set_opus_config", Napi::Function::New(env, set_opus_config));
 
   exports.Set("notify", Napi::Function::New(env, notify));
   exports.Set("notify_xfer", Napi::Function::New(env, notify_xfer));

package/src/pjsip/include/pjsip/sip_transport_ws.h

@@ -0,0 +1,89 @@
+/* 
+ * Copyright (C) 2024 MayamaTakeshi
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA 
+ */
+#ifndef __PJSIP_TRANSPORT_WS_H__
+#define __PJSIP_TRANSPORT_WS_H__
+
+/**
+ * @file sip_transport_ws.h
+ * @brief SIP WebSocket Transport.
+ */
+
+#include <pjsip/sip_transport.h>
+#include <websock.h>
+
+PJ_BEGIN_DECL
+
+/* Dynamic transport type IDs for WS and WSS */
+PJ_DECL_DATA(pjsip_transport_type_e) PJSIP_TRANSPORT_WS;
+PJ_DECL_DATA(pjsip_transport_type_e) PJSIP_TRANSPORT_WSS;
+
+/**
+ * Initialize WebSocket transport module. Must be called once before
+ * any other pjsip_ws_* functions. Registers the WS and WSS transport
+ * types with PJSIP.
+ *
+ * @return PJ_SUCCESS on success.
+ */
+PJ_DECL(pj_status_t) pjsip_ws_transport_init(void);
+
+/**
+ * Start a WebSocket listener (server mode). Creates a WebSocket listener
+ * on the specified address and port. When a WebSocket connection with
+ * SIP sub-protocol is accepted, a new PJSIP transport is automatically
+ * created and registered.
+ *
+ * @param endpt         The SIP endpoint.
+ * @param ws_endpt      The WebSocket endpoint.
+ * @param bind_addr     Address and port to listen on.
+ * @param secure        Use TLS (WSS) if PJ_TRUE, plain WS if PJ_FALSE.
+ * @param p_listener_tp Optional pointer to receive the listener transport.
+ *
+ * @return PJ_SUCCESS on success.
+ */
+PJ_DECL(pj_status_t) pjsip_ws_transport_start(
+    pjsip_endpoint *endpt,
+    pj_websock_endpoint *ws_endpt,
+    const pj_sockaddr *bind_addr,
+    pj_bool_t secure,
+    pjsip_transport **p_listener_tp);
+
+/**
+ * Create a WebSocket connection transport (client mode). Connects to the
+ * specified WebSocket URL and registers a PJSIP transport for it.
+ *
+ * @param endpt         The SIP endpoint.
+ * @param ws_endpt      The WebSocket endpoint.
+ * @param ws_url        WebSocket URL (e.g. "ws://127.0.0.1:8080/sip").
+ * @param hdrs          Additional HTTP headers for the upgrade request
+ *                      (can be NULL).
+ * @param hdr_cnt       Number of headers in hdrs.
+ * @param p_transport   Pointer to receive the created transport.
+ *
+ * @return PJ_SUCCESS on success.
+ */
+PJ_DECL(pj_status_t) pjsip_ws_transport_connect(
+    pjsip_endpoint *endpt,
+    pj_websock_endpoint *ws_endpt,
+    const char *ws_url,
+    const pj_websock_http_hdr *hdrs,
+    int hdr_cnt,
+    pjsip_transport **p_transport);
+
+PJ_END_DECL
+
+#endif  /* __PJSIP_TRANSPORT_WS_H__ */