Jay Desai
2018-10-09 05:35:22 UTC
Hi,
I've a setup of Opensips (with auth) and Freeswitch. Linphone registers to
Opensips server using TLS, I've already set Encryption to DTLS, enabled ICE
added a STUN server, enabled AVPF, However when I originate call from
Freeswitch using media_webrtc=true, linphone is rejecting call with *488
Not acceptable here*. Ideally as per the documentation in this link
<https://wiki.linphone.org/xwiki/wiki/public/view/Lib/Interoperability%20WebRTC/>
it should work.
Looks like Linphone is not able to understand webrtc call and it looking
for a=crypto line in SDP. I've tried this on all the clients(ios,android &
desktop-linux).
I've a setup of Opensips (with auth) and Freeswitch. Linphone registers to
Opensips server using TLS, I've already set Encryption to DTLS, enabled ICE
added a STUN server, enabled AVPF, However when I originate call from
Freeswitch using media_webrtc=true, linphone is rejecting call with *488
Not acceptable here*. Ideally as per the documentation in this link
<https://wiki.linphone.org/xwiki/wiki/public/view/Lib/Interoperability%20WebRTC/>
it should work.
Looks like Linphone is not able to understand webrtc call and it looking
for a=crypto line in SDP. I've tried this on all the clients(ios,android &
desktop-linux).