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

[Benjamin Keith Ford (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-30120?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Benjamin Keith Ford updated ASTERISK-30120:
-------------------------------------------

    Assignee: Neeraj
      Status: Waiting for Feedback  (was: Triage)

# Did this start happening after an upgrade? If so, what version were you on before?
# Do you have any logs or packet traces showing this happening?
# Is this happening with all calls or a specific scenario?

> 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
>            Assignee: 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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