I'm a Software Architect working for a Collaboration Service Provider. One of our solutions is based on the Adobe Flash Media Gateway and I’m currently evaluating Flashphoner RTMP SIP Gateway as a possible replacement candidate. I downloaded the complete RTMP SIP Gateway package and tried the PhoneJS.html example. It works smooth, nice Then I had a look at the RTMP packets using WireShark. I expected to see Speex as the preferred audio codec but instead I see G711. I checked the flashphoner.properties file on the server, it specifies: codecs = speex16,alaw,ulaw Is there anything I'm doing wrong or missing ? Thank You in advance for your answers.