I am using a WebRTC phone to make PSTN calls. When I route the call through an Airtel gateway, both the call connection and audio work properly. However, when I use a Jio network gateway, the call gets connected but RTP packets do not flow, resulting in no audio. I am trying to identify the root cause of this issue and how to resolve it.
You’ll need to examine the ICE negotiation, then, as it is what figures out the route to use for media. Examine the SDP on both sides to look at the ICE candidates and see if logically they should be reachable. Do a packet capture to see if STUN traffic is flowing (ICE uses STUN). If ICE negotiation fails because the two sides can’t reach each other, then media will fail to flow.