[asterisk-bugs] [JIRA] (ASTERISK-30120) RTP Timestamp issue causes clicks & pops on calls routed via alternate Call Platforms

[Neeraj (JIRA)]

Neeraj created ASTERISK-30120:
---------------------------------

             Summary: RTP Timestamp issue causes clicks & pops on calls routed via alternate Call Platforms
                 Key: ASTERISK-30120
                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-30120
             Project: Asterisk
          Issue Type: Bug
      Security Level: None
          Components: Resources/res_rtp_asterisk
    Affects Versions: 16.25.2
         Environment: Centos 7; Centos 8 Streams
            Reporter: Neeraj


Asterisk version: 16.25.2
Problem description:
RTP Timestamp issue causes clicks & pops on calls routed via alternate Call Platforms.

Cause:
This means that a short frame following a typical frame should have a timestamp that is a typical frame offset in time, it is the frame after the short frame that should have a shorter timestamp offset. However this does not seem to be the case currently.

With a shortened packet, the wrong value will be written
static int rtp_raw_write(struct ast_rtp_instance *instance, struct ast_frame *frame, int codec)
{
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
int pred, mark = 0;
unsigned int ms = calc_txstamp(rtp, &frame->delivery);
struct ast_sockaddr remote_address = { {0,} };
int rate = ast_rtp_get_rate(frame->subclass.format) / 1000;
unsigned int seqno;
#ifdef TEST_FRAMEWORK
struct ast_rtp_engine_test test = ast_rtp_instance_get_test(instance);
#endif
if (ast_format_cmp(frame->subclass.format, ast_format_g722) == AST_FORMAT_CMP_EQUAL) {
frame->samples /= 2;
}
if (rtp->sending_digit) {
return 0;
}
#ifdef TEST_FRAMEWORK
if (test && test->send_report) {
test->send_report = 0;
ast_rtcp_write(instance);
return 0;
}
#endif
if (frame->frametype == AST_FRAME_VOICE) {
pred = rtp->lastts + frame->samples;
/ Re-calculate last TS /
rtp->lastts = rtp->lastts + ms * rate;
if (ast_tvzero(frame->delivery)) {
/ If this isn’t an absolute delivery time, Check if it is close to our prediction,
and if so, go with our prediction */
if (abs((int)rtp->lastts - pred) < MAX_TIMESTAMP_SKEW) {
rtp->lastts = pred;
} else {
ast_debug_rtp(3, “(%p) RTP audio difference is %d, ms is %u\n”,
instance, abs((int)rtp->lastts - pred), ms);
mark = 1;
}





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