res_rtp_asterisk.c: Set mark on rtp when timestamp skew is too big

[n2p-litvinov]

We faced with no audio issue when media source changed during the call (e.g. call transfer, audio prompt and then some sip endpoint, etc.). Issue related to big timestamp skew. Solution here is to set Mark when we detect such case.

[gtjoseph]

@n2p-litvinov Is there an issue open for this? If not, you may want to open one. This allows the fix to be highlighted at the top of the changelog.

[gtjoseph]

Please see the following links for proper commit message formatting:
See:
https://docs.asterisk.org/Development/Policies-and-Procedures/Code-Contribution/
https://docs.asterisk.org/Development/Policies-and-Procedures/Commit-Messages/

[github-actions]

REMINDER: If this PR applies to other branches, please add a comment with the appropriate "cherry-pick-to" headers as per the Create a Pull Request process.

If you don't want it cherry-picked, please add a comment with cherry-pick-to: none so we don't keep asking.

If, after adding "cherry-pick-to" comments, you change your mind, please edit the comment to DELETE the header lines and add cherry-pick-to: none.

The currently active branches are now 18, 20, 21, 22 and master.

[n2p-litvinov]

cherry-pick-to: 20
cherry-pick-to: 21
cherry-pick-to: 22

[n2p-litvinov]

Thanks Joseph.
I created an issue. Commit message was corrected. PR was updated.