windows force datatype conversion on uint64_t to switch_size_t for recent change. maybe we should change the structure to be uint64_t as well?

re-mark cur_payload as negotiated when detected as such by parser or the rtp could stop working on session re-invite

FS-7015 #resolve #comment The code was not properly catching the 0.0.0.0 after changing it to work with ICE SDPs because it was looking in the wrong place for the 0.0.0.0

only negotiate codecs that were previously negotiated on re-invites unless its intentional to change codecs

FS-6891 FS-7002 FS-7059 FS-7072 FS-7073 FS-7076 #close #comment All of these bugs are invalidated due to a botched revert -- commit 72c3df5e was supposed to revert commit 1b612fec but it only reverted part of it.  commit 158c1f23 from the other day was erroneously created to work around the result of the botched revert so it is reverted.  The result of this commit should be to bring it back to effectively properly revert 1b612fec and now you can test and open new bugs if necessary.  This includes the segfault in eavesdrop, and any media bug releated reports.  Consider working together and only reporting one new bug if you still encounter any more problems and be sure to include details and make sure the jitter buffer is being used.

FS-7047 #resolve #comment [mod_rayo] arbitrary MRCP headers can now be sent to unimrcp output components

FS-6688: don't update the contact when the original had fs_path, the new one doesnt, and the contact didn't change.  This will fix the normal case of record route from a proxy without breaking normal changing of a contact

FS-7062: [mod_sofia] on redirect, when uri are passed in without <> with multiple uris, automatically add the q= header param in decending order.  This should make 300 Multiple Choices work well with devices that require the q param.  If you would like to specify explicit q-values, please use the syntax of redirect where you specify the entire header using the <>

As per RFC4856, Annex B mode for G.729 is disabled if the SDP contains a
fmtp parameter annexb=no and enabled if annexb=yes or the annexb parameter
is missing.

When responding to an offer mod_sangoma_codec always returns an answer
without a fmtp parameter. Even in response to a SDP offer which contains
one. This has the effect of responding to a request to disable Annex B
with an answer that only Annex B is supported.

With this change mod_sangoma_codec copies the fmtp parameter from the
offer to the answer fixing the problem.

Fixes a bug where the jitter buffer implementation wasn't discarding
old packets if debug mode was disabled.

FS-7045 #resolve #comment [mod_rayo] guarantee that dialed call can be joined when answered event is sent.

This is nothing more than a shameless copy/paste from another mod_say
module, which already had timezone support. It simply checks if the
timezone variable is set and if it contains a valid timezone, then this
timezone will be used when announcing times/dates.

FS-7048 #resolve

FS-7031 #resolve #comment [unimrcp] update library again to pull in upstream fix for --with-sofia-sip=../sofia-sip