Hello, The following is on FreeSWITCH Version 1.5.15b+git~20150126T215733Z~c16f9ec1d9~64bit (git c16f9ec 2015-01-26 21:57:33Z 64bit). The design goal for this configuration is that of a simple transcoding SBC. SIP calls arrive with various supported codecs, SIP calls bridge out on PCMU. No users, no auth, etc. Overall, it seems to work but there is one call flow I'm struggling with. The B-leg of calls are bridged to a PSTN gateway. If the gateway's signaling follows 100, 183, 200, all is well. But if the gateway sends multiple 183s with different RTP ports, there is no audio when the call goes to 200. See the following example: - Gateway signals 183 with SDP indicating audio on port 16384. - Gateway signals 183 with SDP indicating audio on port 16386. - Gateway signals 200 with SDP indicating audio on port 16384 (same as original 183). In the debug I see where it detects the port change from 183 #1 to 183 #2. As such, I hear early media from both until the 200 OK. When the call connects, I see no such port change in the debug, and since it's still listening on the wrong port (from 183 #2), there is no audio. I've seen some older posts where Anthony seemed against even the port change from 183 #1 to 183 #2, yet that seems to work okay today. It just doesn't sense the port change from 183 --> 200. I don't know if this is a feature, bug, or misconfigured option. Thoughts are welcome! - Jeff