I noticed that in the 9.1(1) SDK documentation a new feature was added to the Legacy RTP URI's to select between speakerphone and headset. However, I do not see this feature in the RTP Streaming API. Was this an oversight on my part, something that was left out of the documentation or does this feature not exist? If it does not exist will it in the future? I really could utilize this feature but my application utilizes G722 as the codec becuase it is a higher quality codec and this is only availble in the RTP Streaming API.
From the Documentation:
s = introduced in Release 1.4(4), this parameter specifies where the audio for an XSI call should be played. If s = 1, then the audio for the XSI call will be played to the handset speaker or headset. If s = 0, then the audio for the XSI call will be played to the speaker phone. If s is not present, then the audio for the XSI call is played to the speaker phone.