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Fixed XOR encryption for custom RTP transport modifier tests

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This commit is contained in:
Sylvain Berfini 2015-08-24 16:58:20 +02:00
parent 3f14debcbd
commit e7dd35efa0
1 changed files with 33 additions and 3 deletions
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36
tester/call_tester.c
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@ -4437,17 +4437,34 @@ 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;
rtp_header_t *rtp = (rtp_header_t*)msg->b_rptr;
int i = 0;
unsigned char *src;
int size = 0;
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 msgdsize(msg);
}
// Mediastream can create a mblk_t with only the RTP header and setting the b_cont pointer to the actual RTP content buffer
// In this scenario, the result of rtp_get_payload will be 0, and we won't be able to do our XOR encryption on the payload
// The call to msgpullup will trigger a memcpy of the header and the payload in the same buffer in the msg mblk_t
msgpullup(msg, -1);
// Now that the mblk_t buffer directly contains the header and the payload, we can get the size of the payload and a pointer to it's start (we don't encrypt the RTP header)
size = rtp_get_payload(msg, &src);
// Just for fun, let's do a XOR encryption
for (i = 0; i < size; i++) {
src[i] ^= (unsigned char) XOR_KEY[i % strlen(XOR_KEY)];
}
data->packetSentCount += 1;
/* /!\ DO NOT RETURN 0 or the packet will never leave /!\ */
@ -4459,15 +4476,27 @@ static int rtptm_on_send(RtpTransportModifier *rtptm, mblk_t *msg) {
static int rtptm_on_receive(RtpTransportModifier *rtptm, mblk_t *msg) {
RtpTransportModifierData *data = rtptm->data;
rtp_header_t *rtp = (rtp_header_t*)msg->b_rptr;
int i = 0;
unsigned char *src;
int size = 0;
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 msgdsize(msg);
}
// On the receiving side, there is no need for a msgpullup, the mblk_t contains the header and the payload in the same buffer
// We just ask for the size and a pointer to the payload buffer
size = rtp_get_payload(msg, &src);
// Since we did a XOR encryption on the send side, we have to do it again to decrypt the payload
for (i = 0; i < size; i++) {
src[i] ^= (unsigned char) XOR_KEY[i % strlen(XOR_KEY)];
}
data->packetReceivedCount += 1;
/* /!\ DO NOT RETURN 0 or the packet willbe dropped /!\ */
/* /!\ DO NOT RETURN 0 or the packet will be dropped /!\ */
return msgdsize(msg);
}
@ -4597,7 +4626,7 @@ static void custom_rtp_modifier(bool_t pauseResumeTest, bool_t recordTest) {
player = linphone_call_get_player(call_pauline);
BC_ASSERT_PTR_NOT_NULL(player);
if (player) {
// This will ask marie to play the file
// This will ask pauline 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");
}
@ -4608,6 +4637,7 @@ static void custom_rtp_modifier(bool_t pauseResumeTest, bool_t recordTest) {
end_call(pauline, marie);
// Now we compute a similarity factor between the original file and the one we recorded on the callee side
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");
@ -4635,7 +4665,7 @@ static void custom_rtp_modifier(bool_t pauseResumeTest, bool_t recordTest) {
ms_message("Marie sent %" PRIu64 " RTP packets and received %" PRIu64 " (through our modifier)", data_marie->packetSentCount, data_marie->packetReceivedCount);
ms_message("Pauline sent %" PRIu64 " RTP packets and received %" PRIu64 " (through our modifier)", data_pauline->packetSentCount, data_pauline->packetReceivedCount);
// There will be a few RTP packets sent on marie's side before the call is ended at pauline's request, so we need the threshold
BC_ASSERT_TRUE(MAX(data_pauline->packetReceivedCount, data_marie->packetSentCount) - MIN(data_pauline->packetReceivedCount, data_marie->packetSentCount) < 50);
BC_ASSERT_TRUE(data_marie->packetSentCount - data_pauline->packetReceivedCount < 50);
BC_ASSERT_TRUE(data_marie->packetReceivedCount == data_pauline->packetSentCount);
// At this point, we know each packet that has been processed in the send callback of our RTP modifier also go through the recv callback of the remote.