sip-lab 1.33.0 → 1.34.1

package/DEV.md

@@ -54,7 +54,7 @@ However the above will build the addon to run on the current OS.
 
 Instead we will force the build on debian11 (using docker) using prebuildify-cross. So do this instead:
 
-Make sure you have the docker image built.
+Make sure you have the docker image built (the image must be rebuilt whenever we update build_deps.sh)
 
 cd docker-images/debian11/
 ./build_image.sh

package/build_deps.sh

@@ -39,7 +39,7 @@ then
     git clone https://github.com/MayamaTakeshi/bcg729
     cd bcg729
     git checkout faaa895862165acde6df8add722ba4f85a25007d
-    cmake . 
+    cmake -DCMAKE_BUILD_TYPE=RelWithDebInfo .
     make
     mkdir -p lib
     cp -f src/libbcg729.a lib
@@ -70,6 +70,8 @@ EOF
 #define PJSUA_MAX_ACC (20000)
 #define PJ_IOQUEUE_MAX_HANDLES (1024)
 #define PJSUA_MAX_CALLS (20000)
+
+#define PJMEDIA_HAS_OPUS_CODEC 1
 EOF
     make dep && make clean && make
 fi

package/package.json

@@ -1,6 +1,6 @@
 {
   "name": "sip-lab",
-  "version": "1.33.0",
+  "version": "1.34.1",
   "description": "",
   "main": "index.js",
   "engines": {

package/samples/g729.js

@@ -77,6 +77,23 @@ async function test() {
         },
     ], 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/yosemitesam.wav', end_of_file_event: true, no_loop: true})
+    sip.call.start_play_wav(ic.id, {file: 'samples/artifacts/yosemitesam.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,
+        },
+    ], 3000)
+
     sip.call.reinvite(oc.id)
 
     await z.wait([
@@ -173,6 +190,9 @@ async function test() {
     assert(oc_stat.CodecInfo == 'G729/8000/1')
     assert(ic_stat.CodecInfo == 'G729/8000/1')
 
+    sip.call.stop_record_wav(oc.id)
+    sip.call.stop_record_wav(ic.id)
+
     sip.call.terminate(oc.id)
 
     await z.wait([

package/samples/opus.narrowband.js

@@ -0,0 +1,236 @@
+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.store.call_id,
+        sip_call_id: z.store.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/yosemitesam.wav', end_of_file_event: true, no_loop: true})
+    sip.call.start_play_wav(ic.id, {file: 'samples/artifacts/yosemitesam.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,
+        },
+    ], 3000)
+
+    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.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/src/sip.cpp

@@ -1558,6 +1558,29 @@ int __pjw_init() {
     addon_log(L_DBG, "pjmedia_codec_opus_init failed\n");
     return 1;
   }
+  pjmedia_codec_param param;
+  pjmedia_codec_opus_config opus_cfg;
+
+  pjmedia_codec_mgr *codec_mgr;
+  codec_mgr = pjmedia_endpt_get_codec_mgr(g_med_endpt);
+
+  unsigned cnt = 1;
+  const pjmedia_codec_info *pci;
+
+  pj_str_t codec_id = pj_str((char*)"opus/48000/2");
+  pjmedia_codec_mgr_find_codecs_by_id(codec_mgr, &codec_id, &cnt, &pci, NULL);
+  pjmedia_codec_mgr_get_default_param(codec_mgr, pci, &param);
+  pjmedia_codec_opus_get_config(&opus_cfg);
+
+  opus_cfg.sample_rate = 8000;
+  opus_cfg.channel_cnt = 1;
+  opus_cfg.bit_rate = 8000;
+  status = pjmedia_codec_opus_set_default_param(&opus_cfg, &param);
+  if(status != PJ_SUCCESS)
+  {
+      addon_log(L_DBG, "pjmedia_codec_opus_set_default_param failed\n");
+      return 1;
+  }
 #endif
 
 #if defined(PJMEDIA_HAS_SRTP) && (PJMEDIA_HAS_SRTP != 0)