TT#184200 actualization of 'invite_reinvite_caller_transcoding' and 'invite_reinvite_transcoding'

We need to actualize scenarios when checking usage fmtp description in SDP bodies,
due to late changes in the RTPEngine behavior
(when processing offer/answer with/without fmtp for the 101th event).

Change-Id: I47f97cba69345fb0fdce777831e7f23cc37a79b3
(cherry picked from commit 25307b3146)

scenarios/invite_reinvite_caller_transcoding/sipp_scenario00_test.yml.tt2

@@ -76,7 +76,6 @@ messages:
   - 'a=rtpmap:18 G729/8000'
   - 'a=fmtp:18 annexb=no'
   - 'a=rtpmap:101 telephone-event/8000'
-  - 'a=fmtp:101 0-16'
   - 'a=sendrecv'
   - 'a=rtcp:\d+'
   - 'a=ptime:20'

scenarios/invite_reinvite_transcoding/sipp_scenario_responder00_test.yml.tt2

@@ -58,7 +58,6 @@ messages:
   - 'a=rtpmap:18 G729/8000'
   - 'a=fmtp:18 annexb=no'
   - 'a=rtpmap:101 telephone-event/8000'
-  - 'a=fmtp:101 0-16'
   - 'a=sendrecv'
   - 'a=rtcp:\d+'
   - 'a=silenceSupp:off - - - -'