Freeswitch solution rtmp to sip Gateway

This article from csdn ucser, http://blog.csdn.net/perfectpdl reprinted to indicate the source, thank you!

I have created a freeswitch learning and communication group, 45211986. welcome to join.

Freeswtich can be used as the rtmp and SIP gateway of the Streaming Media Protocol. It can communicate with the SIP video phone through flash in a web browser. This function can be used on the browser side for similar click2call or online video communication functions.

1. make mod_rtmp install2. configure FS to load the rtmp module modues. conf. XML removes the comment of the mod_rtmp module. 3. load module fs_cliload mod_rtmp4. deploy the client to Apache [root @ Union flex] # cp-RF flex // var/www/html/5. configure the client connection flash URL: rtmp_url: 'rtmp: // fs_ip_adress/phone' 6. run the client http: // fs_ip_adress/flex/freeswitch.html 7. the rtmp module uses the same account as the SIP module. users in the directory can directly use it. 8. Configure the SIP client to communicate with each other.