Hi all.
Clearly no expert
I am trying to route SIP calls via Linphone/MicroSIP -> Opensips/RTPEngine -> Freeswitch -> PSTN.
All webrtc calls work fine. SIP calls directly to PSTN also work fine. Phones registered directly on Freeswitch are ok too. However, SIP calls via Freeswitch to PSTN don't have any audio. Following is the SDP exchange at Freeswitch. This (to me atleast) appears to be an RTPEngine issue. Any thoughts please?
Edit: Just fixed (firewall ports issue!)
Clearly no expert
I am trying to route SIP calls via Linphone/MicroSIP -> Opensips/RTPEngine -> Freeswitch -> PSTN.
All webrtc calls work fine. SIP calls directly to PSTN also work fine. Phones registered directly on Freeswitch are ok too. However, SIP calls via Freeswitch to PSTN don't have any audio. Following is the SDP exchange at Freeswitch. This (to me atleast) appears to be an RTPEngine issue. Any thoughts please?
Edit: Just fixed (firewall ports issue!)
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