Adding negotiation of speech codecs between GSM and SIP when using rtp-bridge

Adding negotiation of speech codecs between GSM and SIP when using rtp-bridge

Since LCR does not put hands on any RTP frame when directly bridged between
OpenBSC and SIP, it will now allow all speech codecs that are commonly supported
by MS and remote SIP endpoint.

It must be noted that OpenBSC must support forwarding the codec types that
MS and remote SIP endpoints support.

Currently LCR negotiates the following codecs for GSM:
- Full Rate
- EFR
- AMR
- Half Rate