PJSIP configurations ( no audio between local endpoints & SIP registrations rejected with 403 forbidden error)

Good day everyone,

I’ve got a couple of issues (3 part question) with one of my Labs (PJSIP configuration) :

Scenario 1:

I have two labs and have deployed servers with mirroring PJSIP configurations; however my results are night v day.

Problem 1: No audio between local PJSIP endpoints A & B connected to PBX1. ( SIP signaling using port 5160 / method=UDP ) - using port 5160 to prevent conflicts with working platform that uses port 5060.

  • In this scenario, I grabbed packets from the server and see the SIP signaling passed and processed as normal ( Endpoint A to PBX1 to Endpoint B & vice versa); however, when audio is generated, I see RTP packets reaching the PBX1 interface from the source endpoint (A), but is not forwarded to the remote endpoint (B) and vice versa ( DIDs ie. 2000 and 2001) . I was able to replicate these results on a second serve in Lab 1. ( I see the RTP be relayed to the correct negotiated IPs and Ports within SDP but the server fails to relay the Audio to the 2nd leg of the call)

  • however, in my Home Lab (Lab 2) the replicated configurations and scenarios work like a champ.

it seemed almost similar to an issue ( below), reported a cple of weeks ago, but not apples v apples - does anyone have any suggestions offhand? -> can forward configs l8r

"# [No Audio on Public-facing FreePBX 15 and Asterisk 16 on CentOS 7] "

  • Scenario 2 : PJSIP trunk registrations fail over the internet but the Trunks remain online

  • 2nd issue : PJSIP trunk registrations are rejected with 403’s but remain online and work ( atleast for my home lab) Is this expected behavior for PJSIP trunks? Also what PJSIP trunk configurations would be deemed the most secure over the internet?


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