Added a SDP hack for Firefox to replace the 0.0.0.0 in the c line

In Firefox 34+ they have decided to set the "c" line in the SDP to 0.0.0.0 and in legacy SIP an IP of 0.0.0.0 means "hold the call". Freeswitch interprets this hold by not sending audio back. My patch makes replaces the 0.0.0.0 with 127.0.0.1 which causes Freeswitch to handle the request appropriately.

It's possible that Freeswitch or SIP.js will fix the problem at some point on their own ends, but this is my hack to make sure that WebRTC works when Firefox 34 is released.