[Freeswitch-users] Audio issues when FS as PSTN/SBC Gateway for OpenSIPS

[Deon Vermeulen]

Hi List


My Current setup is as follows.

PSTN -- Sangoma A500 bri -- FS -- OpenSIPS

DID: bridges called number to ext at opensips

My Phones are located on our office private LAN, while FS and OpenSIPS are both on Public Network with Public IPs.
There is NAT between the IP Phones and OpenSIPS/FS.
I am using non standard SIP ports as well as not defined any rules for RTP on the firewall.
I have a many to one NAT scenario on the Firewall.


OpenSIPS is setup to not Proxy any media between the phones and FS.
Media flows directly between FS and phones.

When I make calls to PSTN, audio is Crystal clear on both sides of the call.

When I receive calls from the PSTN I sometimes have audio, sometimes none and most of the time one way (to PSTN).

I have done quite a lot of sip trace debugging in fs_cli and I really can't see any problems in the traces.
I have also enabled inbound-proxy-media in all sip profiles, just to make sure I don't miss it somewhere in the call flow.


I have been digging around in the wiki, forums as well as some archived mails wrt FS and NAT, but I don't 100% understand how FS handles NAT by default in the External.xml profile.
I'm also not 100% sure how FS will handle this type of scenario as no Phones are registered to it and it also has to pass SIP and SDP info with OpenSIPS and not directly with the Phones.


Any advise and or guidance will really be appreciated.


Thank you very much.

Kind Regards
Deon Vermeulen