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

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-30120?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=259780#comment-259780 ] 

Asterisk Team commented on ASTERISK-30120:
------------------------------------------

Suspended due to lack of activity. This issue will be automatically re-opened if the reporter posts a comment. If you are not the reporter and would like this re-opened please create a new issue instead. If the new issue is related to this one a link will be created during the triage process. Further information on issue tracker usage can be found in the Asterisk Issue Guidlines [1].

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines

> 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
> {code:title=res_rtp_asterisk.c}
> 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;
>             }
> {code}



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