res_pjsip_sdp_rtp: Fix intermittent call drop on reINVITE sdp change

res_pjsip_sdp_rtp: Fix intermittent call drop on reINVITE sdp change

A reINVITE that changed the offered codec set (e.g. ulaw+alaw -> alaw
only) could lead to call teardown when a bridge write still using the
old format occurred during the re-negotiation.

This change avoids the termination by making the following changes.

First, Asterisk now holds the lock across both the tx payload update
and the channel format update so they commit atomically.

Second, when processing the sdp on the new incoming offer, merge the
new offer with the old until the negotiation is complete.

Also adds unit tests for the sdp merge

Fixes: #2059