Here you can find a port for Asterisk trunk (rev. 377802) of this patch: http://lists.digium.com/pipermail/asterisk-dev/2011-September/051262.html.
This patch permits to set the codec of an outgoing multicast RTP stream via the variable MULTICAST_RTP_CODEC:

exten => 999,1,Answer()  
exten => 999,n,Set(MULTICAST_RTP_CODEC=alaw ...

Starting from Asterisk 1.8 you can send multicast rtp streams using the MulticastRTP channel driver. There is an open issue that breaks outgoing RTP if the source channel doesn't contains timing informations (Eg. playing an audio file with Dial(MulticastRTP/basic/239.255.255.245:5555,,A(my-announce ...