Copy URI params from Refer-To header into custom header in subsequent INVITE

sip_h_X-FS-Refer-Params is set to show up in the INVITE as X-FS-Refer-Params

According to https://code.google.com/p/webrtc/issues/detail?id=2768 ; The Chrome WebRTC engine reserves payload 98 and 99, IKR?  So, to avoid taking a nasty spill down the stairs and subjecting ourselves to further school absences, we'll just start our payload space at 102 when making WebRTC calls.......

add a way to tell mod_conference when the rate of the channel has changed due to a codec change so it can reset the resampler and codecs internally

feed all packets to jitterbuffer when enabled to absorb bursts and improve smoothing and delay protection

If the input started with 'sip:sips:' it would have been incorrectly
parsed.

add optional rtp_secure_media_suites variable clobbered by rtp_secure_media with mandatory|optional:<suites>

rtp_secure_media=mandatory
rtp_secure_media=optional
rtp_secure_media=mandatory:AES_CM_256_HMAC_SHA1_80,AES_CM_256_HMAC_SHA1_32
rtp_secure_media=optional:AES_CM_256_HMAC_SHA1_80
rtp_secure_media=forbidden

true implies mandatory
false implies forbidden
not set implies optional

rtp_secure_media_inbound or rtp_secure_media_outbound take precedence and are treated the same way based on leg direction

This generates an identical list of cipher suites, but this commit
restates the cipher spec to be more similar to the way we state it
elsewhere.

Previously we disallowed anonymous Diffie-Hellman, but there are other
kinds of null-authentication TLS suites.  In particular, disallowing
AECDH is important now that we support elliptic-curve Diffie-Hellman.