sip-lab 1.13.0 → 1.13.1

package/README.md

@@ -10,19 +10,22 @@ It permits to:
   - send and receive DTMF inband/RFC2833/INFO.
   - play/record wav file on a call
   - send/receive fax (T.30 only)
-  - send/receive MRCPv2 messages (experimental)
+  - send/receive MRCPv2 messages (TCP only, no TLS)
 
 TODO:
-  - add suport for T.38 fax
-  - add support for WebRTC
   - add support for video playing/recording from/to file
+  - add support for T.38 fax
+  - add support for WebSocket
+  - add support for WebRTC
+  - add support for SRTP
+  - add support for MSRP
 
 ### Installation
 
 This is a node.js addon and it is known to work on Debian 11, Debian 10, Ubuntu 22.04 and Ubuntu 20.04.
 It is distributed with prebuild binaries for node.js 15.0.0 and above (but built for Debian 11. For other Debian versions or for Ubuntu a local built of the addon will be executed. Being the case, be patient as the build process will take several minutes to complete).
 
-To install it, first install some dependencies (you might not need them if your are on Debian 11).
+To install it, first install some build dependencies (you might not need them if your are on Debian 11).
 ```
 apt install build-essential automake autoconf libtool libspeex-dev libopus-dev libsdl2-dev libavdevice-dev libswscale-dev libv4l-dev libopencore-amrnb-dev libopencore-amrwb-dev libvo-amrwbenc-dev libvo-amrwbenc-dev libboost-dev libtiff-dev libpcap-dev libssl-dev uuid-dev cmake
 ```

package/binding.gyp

@@ -44,7 +44,6 @@
                 '-L ../3rdParty/pjproject/third_party/lib',
                 '-l pjnath-x86_64-unknown-linux-gnu',
                 '-l ilbccodec-x86_64-unknown-linux-gnu',
-                '-l srtp-x86_64-unknown-linux-gnu',
                 '-l webrtc-x86_64-unknown-linux-gnu',
                 '-l yuv-x86_64-unknown-linux-gnu',
                 '-l speex-x86_64-unknown-linux-gnu',
@@ -87,6 +86,7 @@
                 '-lopencore-amrwb',
                 '-lvo-amrwbenc',
                 '-lspeex',
+                '-l srtp-x86_64-unknown-linux-gnu',
               ],
           },
        },

package/build_deps.sh

@@ -54,7 +54,8 @@ then
     #git checkout de3d744c2e1188b59bb907b6ee32ef83740ebc64
     #git checkout 33a3c9e0a5eb84426edef05a9aa98af17d8011c3 # required for bcg729
     #git checkout 797088ed133c98492519b7d042b75735f6f9388c # updated as part of #21
-    git checkout 651df5b50129b7c5a5feec8336dda4468d53d2b0 # updated to latest to see of crash issues improve
+    #git checkout 651df5b50129b7c5a5feec8336dda4468d53d2b0 # updated to latest to see of crash issues improve
+    git checkout 043926a5846963a2c99378e8daa495230923eaab # update to try to solve ##49 (but issue remains)
 
     cat > user.mak <<EOF
     export CFLAGS += -fPIC -g
@@ -64,7 +65,6 @@ EOF
     sed -i -r 's/BCG729_LIBS="-lbcg729"/BCG729_LIBS=''/' aconfigure
     LIBS=`pwd`/../bcg729/src/libbcg729.a ./configure --with-bcg729=`pwd`/../bcg729
     cat > pjlib/include/pj/config_site.h <<EOF
-#define PJMEDIA_HAS_SRTP  0
 EOF
     make dep && make clean && make
 fi

package/package.json

@@ -1,6 +1,6 @@
 {
   "name": "sip-lab",
-  "version": "1.13.0",
+  "version": "1.13.1",
   "description": "",
   "main": "index.js",
   "engines": {

package/samples/reinvite_audio_audio.js

@@ -130,7 +130,7 @@ async function test() {
             mode: 1,
             media_id: 1
         },
-    ], 1500)
+    ], 2000)
 
     sip.call.reinvite(oc.id, {media: 'audio,audio'})
 

package/src/sip.cpp

@@ -433,7 +433,7 @@ struct AsockUserData {
   MediaEndpoint *media_endpt;
   Call *call;
   char buf[MAX_TCP_DATA];
-  char len;
+  pj_size_t len;
 };
 
 bool init_media_ports(Call *c, AudioEndpoint *ae, unsigned sampling_rate,
@@ -841,18 +841,20 @@ static pj_bool_t on_data_read(pj_activesock_t *asock, void *data,
   AsockUserData *ud = (AsockUserData*)pj_activesock_get_user_data(asock);
   if(!ud) return PJ_FALSE;
 
-  printf("%.*s\n", size, data);
   if (size == 0) {
     // TODO: destroy the activesock.
     return PJ_FALSE;
   }
 
-  assert(size + ud->len < MAX_TCP_DATA);
+  assert(ud->len >= 0);
+
+  assert(size >= 0);
+  assert(size + ud->len + 1 < MAX_TCP_DATA);
 
   memcpy(&ud->buf[ud->len], data, size);
-  ud->len = size + ud->len;
+  ud->len = ud->len + size;
   ud->buf[ud->len] = '\0';
-  
+
   char *sep = strstr(ud->buf, "\r\n\r\n");
   if(!sep) {
     // msg incomplete
@@ -877,8 +879,10 @@ static pj_bool_t on_data_read(pj_activesock_t *asock, void *data,
     num_str[len] = '\0';
     int body_len = atoi(num_str);
 
-    if(sep+4+body_len < ud->buf+ud->len) {
-      // msg incomplete
+    assert(body_len > 0 && body_len < 4096);
+
+    if(ud->buf+ud->len < sep+4+body_len) {
+      printf("tcp data: msg incomplete %i %i\n", ud->buf+ud->len, sep+4+body_len);
       *remainder = 0;
       return PJ_TRUE;
     }
@@ -3294,7 +3298,7 @@ int pjw_call_start_record_wav(long call_id, const char *json) {
   pj_status_t status;
   pjmedia_port *stream_port;
 
-  unsigned media_id;
+  unsigned media_id = 0;
 
   MediaEndpoint *me;
   AudioEndpoint *ae;
@@ -3399,7 +3403,7 @@ int pjw_call_start_play_wav(long call_id, const char *json) {
   AudioEndpoint *ae;
   int ae_count;
 
-  unsigned media_id;
+  unsigned media_id = 0;
 
   char *file;
 
@@ -3527,7 +3531,7 @@ int pjw_call_stop_play_wav(long call_id, const char *json) {
   int ae_count;
   int res;
 
-  unsigned media_id;
+  unsigned media_id = 0;
 
   char buffer[MAX_JSON_INPUT];
 
@@ -3624,7 +3628,7 @@ int pjw_call_stop_record_wav(long call_id, const char *json) {
   int ae_count;
   int res;
 
-  unsigned media_id;
+  unsigned media_id = 0;
 
   char buffer[MAX_JSON_INPUT];
 
@@ -3703,7 +3707,7 @@ int pjw_call_start_fax(long call_id, const char *json) {
   AudioEndpoint *ae;
   int ae_count;
 
-  unsigned media_id;
+  unsigned media_id = 0;
 
   char buffer[MAX_JSON_INPUT];
 
@@ -3841,7 +3845,7 @@ int pjw_call_stop_fax(long call_id, const char *json) {
   int ae_count;
   int res;
 
-  unsigned media_id;
+  unsigned media_id = 0;
 
   char buffer[MAX_JSON_INPUT];
 
@@ -3924,7 +3928,7 @@ int pjw_call_get_stream_stat(long call_id, const char *json, char *out_stats) {
   AudioEndpoint *ae;
   VideoEndpoint *ve;
 
-  unsigned media_id;
+  unsigned media_id = 0;
 
   pjmedia_stream *med_stream = NULL;
 
@@ -7601,7 +7605,7 @@ int pjw_call_send_tcp_msg(long call_id, const char *json) {
   MrcpEndpoint *mrcp_endpt;
   int res;
 
-  unsigned media_id;
+  unsigned media_id = 0;
 
   char buffer[MAX_JSON_INPUT];