From 7cbb44281c1ab12d90b3b712d2a9087df66207d2 Mon Sep 17 00:00:00 2001
From: Sylvain Berfini <sylvain.berfini@belledonne-communications.com>
Date: Fri, 21 Aug 2015 17:46:51 +0200
Subject: [PATCH] Added record test with custom RTP transport modifier but
removed XOR encryption temporarily (doesn't work really great, have to rework
it)
---
tester/call_tester.c | 109 +++++++++++++++++++++++++++----------------
1 file changed, 68 insertions(+), 41 deletions(-)
diff --git a/tester/call_tester.c b/tester/call_tester.c
index 3c234b6402..1291d76129 100644
--- a/tester/call_tester.c
+++ b/tester/call_tester.c
@@ -4437,30 +4437,19 @@ typedef struct _RtpTransportModifierData {
uint64_t packetReceivedCount;
} RtpTransportModifierData;
-const char *XOR_KEY = "BELLEDONNECOMMUNICATIONS";
-
// Callback called when a packet is on it's way to be sent
// This is where we can do some changes on it, like encrypt it
static int rtptm_on_send(RtpTransportModifier *rtptm, mblk_t *msg) {
RtpTransportModifierData *data = rtptm->data;
- int i = 0;
- int size = msg->b_wptr - msg->b_rptr;
- unsigned char *src = msg->b_rptr;
rtp_header_t *rtp = (rtp_header_t*)msg->b_rptr;
if (rtp->version == 0) {
// This is probably a STUN packet, so don't count it (oRTP won't) and don't encrypt it either
- return size;
+ return msgdsize(msg);
}
data->packetSentCount += 1;
- // Just for fun, let's do a XOR encryption
- for (i = 0; i < size; i++) {
- src[0] ^= XOR_KEY[i % strlen(XOR_KEY)];
- src++;
- }
-
/* /!\ DO NOT RETURN 0 or the packet will never leave /!\ */
return msgdsize(msg);
}
@@ -4469,24 +4458,15 @@ static int rtptm_on_send(RtpTransportModifier *rtptm, mblk_t *msg) {
// This is where we can do some changes on it, like decrypt it
static int rtptm_on_receive(RtpTransportModifier *rtptm, mblk_t *msg) {
RtpTransportModifierData *data = rtptm->data;
- int i = 0;
- int size = msg->b_wptr - msg->b_rptr;
- unsigned char *src = msg->b_rptr;
rtp_header_t *rtp = (rtp_header_t*)msg->b_rptr;
if (rtp->version == 0) {
// This is probably a STUN packet, so don't count it (oRTP won't) and don't decrypt it either
- return size;
+ return msgdsize(msg);
}
data->packetReceivedCount += 1;
- // Do the XOR decryption
- for (i = 0; i < size; i++) {
- src[0] ^= XOR_KEY[i % strlen(XOR_KEY)];
- src++;
- }
-
/* /!\ DO NOT RETURN 0 or the packet willbe dropped /!\ */
return msgdsize(msg);
}
@@ -4549,43 +4529,61 @@ static void custom_rtp_modifier(bool_t pauseResumeTest, bool_t recordTest) {
RtpTransportModifier *rtptm_pauline = NULL;
RtpTransportModifierData *data_marie = NULL;
RtpTransportModifierData *data_pauline = NULL;
+ // The following are only used for the record test
+ LinphonePlayer *player;
+ char *hellopath = bc_tester_res("sounds/ahbahouaismaisbon.wav"); // File to be played
+ char *recordpath = create_filepath(bc_tester_get_writable_dir_prefix(), "record-call_with_file_player", "wav"); // File to record the received sound
+ double similar = 1; // The factor of similarity between the played file and the one recorded
+ const double threshold = 0.9; // Minimum similarity value to consider the record file equal to the one sent
// We create a new vtable to listen only to the call state changes, in order to plug our RTP Transport Modifier when the call will be established
v_table = linphone_core_v_table_new();
- v_table->call_state_changed=call_state_changed_4;
+ v_table->call_state_changed = call_state_changed_4;
linphone_core_add_listener(pauline->lc,v_table);
v_table = linphone_core_v_table_new();
- v_table->call_state_changed=call_state_changed_4;
+ v_table->call_state_changed = call_state_changed_4;
linphone_core_add_listener(marie->lc,v_table);
+
+ if (recordTest) { // When we do the record test, we need a file player to play the content of a sound file
+ /*make sure the record file doesn't already exists, otherwise this test will append new samples to it*/
+ unlink(recordpath);
+
+ linphone_core_use_files(pauline->lc,TRUE);
+ linphone_core_set_play_file(pauline->lc,NULL);
+ linphone_core_set_record_file(pauline->lc,NULL);
+
+ /*callee is recording and plays file*/
+ linphone_core_use_files(marie->lc,TRUE);
+ linphone_core_set_play_file(marie->lc,NULL);
+ linphone_core_set_record_file(marie->lc,recordpath);
+ }
// Now the the call should be running (call state StreamsRunning)
BC_ASSERT_TRUE((call_ok=call(pauline,marie)));
if (!call_ok) goto end;
-
- // This only wait for 3 seconds in order to generate traffic for the test
- wait_for_until(pauline->lc, marie->lc, NULL, 5, 3000);
// Ref the call to keep the pointer valid even after the call is release
call_pauline = linphone_call_ref(linphone_core_get_current_call(pauline->lc));
- rtptm_pauline = (RtpTransportModifier *)call_pauline->user_data;
call_marie = linphone_call_ref(linphone_core_get_current_call(marie->lc));
- rtptm_marie = (RtpTransportModifier *)call_marie->user_data;
// This is for the pause/resume test, we don't do it in the call record test to be able to check the recorded call matches the file played
if (pauseResumeTest) {
+ // This only wait for 3 seconds in order to generate traffic for the test
+ wait_for_until(pauline->lc, marie->lc, NULL, 5, 3000);
+
linphone_core_pause_call(pauline->lc,call_pauline);
- BC_ASSERT_TRUE(wait_for(pauline->lc,marie->lc,&pauline->stat.number_of_LinphoneCallPausing,1));
- BC_ASSERT_TRUE(wait_for(pauline->lc,marie->lc,&marie->stat.number_of_LinphoneCallPausedByRemote,1));
- BC_ASSERT_TRUE(wait_for(pauline->lc,marie->lc,&pauline->stat.number_of_LinphoneCallPaused,1));
+ BC_ASSERT_TRUE(wait_for(pauline->lc, marie->lc, &pauline->stat.number_of_LinphoneCallPausing, 1));
+ BC_ASSERT_TRUE(wait_for(pauline->lc, marie->lc, &marie->stat.number_of_LinphoneCallPausedByRemote, 1));
+ BC_ASSERT_TRUE(wait_for(pauline->lc, marie->lc, &pauline->stat.number_of_LinphoneCallPaused, 1));
/*stay in pause a little while in order to generate traffic*/
wait_for_until(pauline->lc, marie->lc, NULL, 5, 2000);
linphone_core_resume_call(pauline->lc,call_pauline);
- BC_ASSERT_TRUE(wait_for(pauline->lc,marie->lc,&pauline->stat.number_of_LinphoneCallStreamsRunning,2));
- BC_ASSERT_TRUE(wait_for(pauline->lc,marie->lc,&marie->stat.number_of_LinphoneCallStreamsRunning,2));
+ BC_ASSERT_TRUE(wait_for(pauline->lc, marie->lc, &pauline->stat.number_of_LinphoneCallStreamsRunning, 2));
+ BC_ASSERT_TRUE(wait_for(pauline->lc, marie->lc, &marie->stat.number_of_LinphoneCallStreamsRunning, 2));
/*same here: wait a while for a bit of a traffic, we need to receive a RTCP packet*/
wait_for_until(pauline->lc, marie->lc, NULL, 5, 5000);
@@ -4593,16 +4591,42 @@ static void custom_rtp_modifier(bool_t pauseResumeTest, bool_t recordTest) {
/*since RTCP streams are reset when call is paused/resumed, there should be no loss at all*/
stats = rtp_session_get_stats(call_pauline->sessions->rtp_session);
BC_ASSERT_EQUAL(stats->cum_packet_loss, 0, int, "%d");
- }
- // We termine the call and check the stats to see if the call is correctly ended on both sides
- linphone_core_terminate_all_calls(pauline->lc);
- BC_ASSERT_TRUE(wait_for(pauline->lc,marie->lc,&pauline->stat.number_of_LinphoneCallEnd,1));
- BC_ASSERT_TRUE(wait_for(pauline->lc,marie->lc,&marie->stat.number_of_LinphoneCallEnd,1));
+ end_call(pauline, marie);
+ } else if (recordTest) {
+ player = linphone_call_get_player(call_pauline);
+ BC_ASSERT_PTR_NOT_NULL(player);
+ if (player) {
+ // This will ask marie to play the file
+ BC_ASSERT_EQUAL(linphone_player_open(player, hellopath, on_eof, pauline),0, int, "%d");
+ BC_ASSERT_EQUAL(linphone_player_start(player), 0, int, "%d");
+ }
+ /* This assert should be modified to be at least as long as the WAV file */
+ BC_ASSERT_TRUE(wait_for_until(pauline->lc, marie->lc, &pauline->stat.number_of_player_eof, 1, 10000));
+ /*wait one second more for transmission to be fully ended (transmission time + jitter buffer)*/
+ wait_for_until(pauline->lc, marie->lc, NULL, 0, 1000);
+
+ end_call(pauline, marie);
+
+ BC_ASSERT_EQUAL(ms_audio_diff(hellopath, recordpath, &similar, audio_cmp_max_shift, NULL, NULL), 0, int, "%d");
+
+ BC_ASSERT_GREATER(similar, threshold, double, "%g");
+ BC_ASSERT_LOWER(similar, 1.0, double, "%g");
+ if (similar >= threshold && similar <= 1.0) {
+ // If the similarity value is between perfect (1) and our threshold (0.9), then we delete the file used for the record
+ remove(recordpath);
+ }
+ } else {
+ // This only wait for 3 seconds in order to generate traffic for the test
+ wait_for_until(pauline->lc, marie->lc, NULL, 5, 3000);
+
+ // We termine the call and check the stats to see if the call is correctly ended on both sides
+ end_call(pauline, marie);
+ }
// Now we can go fetch our custom structure and check the number of packets sent/received is the same on both sides
- BC_ASSERT_PTR_NOT_NULL(rtptm_marie);
- BC_ASSERT_PTR_NOT_NULL(rtptm_pauline);
+ rtptm_marie = (RtpTransportModifier *)call_marie->user_data;
+ rtptm_pauline = (RtpTransportModifier *)call_pauline->user_data;
data_marie = (RtpTransportModifierData *)rtptm_marie->data;
data_pauline = (RtpTransportModifierData *)rtptm_pauline->data;
@@ -4651,6 +4675,9 @@ end:
// The test is finished, the linphone core are no longer needed, we can safely free them
linphone_core_manager_destroy(marie);
linphone_core_manager_destroy(pauline);
+
+ ms_free(recordpath);
+ ms_free(hellopath);
}
static void call_with_custom_rtp_modifier(void) {
--
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