RTMP SIP Gateway and Speex

Silvio Huckel

New Member
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.


Staff member
Our support team says they have already sent this reply by email. Please check.

Try to add:
force_local_audio_codec = speex16
We love g.711 because this codec is supported everywhere(on SIP) and does not require transcoding.
By the way, you can consider our product WCS4 as a replacement of Adobe Flash Media Gateway.
Recently we have implemented high quality SIP as RTMP support for H.264 and Speex, G.711 codecs in WCS4 with REST API.
So you can call to any SIP destination, fetch SIP/RTP stream and forward this stream to Adobe FMS or other RTMP CDN.
I have attached technical description of the SIP-RTMP delivery usecase. The case is stable and load tested.