WebRTC call fails with INCOMPATIBLE_DESTINATION / 488 Not Acceptable Here (ICE/DTLS candidate issue)

[harikrishnanev-afk]

Description

I am trying to use FreeSWITCH with a WebRTC client (React Native + JsSIP + react-native-webrtc).

• SIP registration over WebSocket works correctly
• User shows as registered in sofia status profile internal reg
• But calls fail during media negotiation with errors like:

INCOMPATIBLE_DESTINATION
488 Not Acceptable Here
no suitable candidates found
Incompatible SDP
DTLS handshake starts but never completes

So signaling is fine, but media (ICE/DTLS/RTP) always fails.

---

Environment

• FreeSWITCH: 1.10.12-release
• OS: Ubuntu
• Transport: SIP over WebSocket (ws)
• Client: React Native + JsSIP 3.11.1 + react-native-webrtc
• Network: Both server and client in same private network (10.10.4.x)

---

What Works

• SIP registration works:

  User: 1000@xxx.xxx.xx.xx
  Status: Registered(WS-NAT)
  Agent: JsSIP 3.11.1
  

• Incoming INVITE reaches FreeSWITCH

• Dialplan matches correctly

• Call enters Pre-Answer state

---

What Fails

During call setup, logs show:

Activating Audio ICE
Activating RTP audio ICE: ...
Activate RTP/RTCP audio DTLS client
Changing audio DTLS state from OFF to HANDSHAKE
...
no suitable candidates found
Hangup ... cause: INCOMPATIBLE_DESTINATION

Or sometimes:

488 Not Acceptable Here
Reason: INCOMPATIBLE_DESTINATION

So FreeSWITCH cannot accept the WebRTC SDP / ICE candidates.

---

SIP Profile Configuration (internal.xml)

<param name="webrtc" value="true"/>
<param name="rtcp-mux" value="true"/>
<param name="disable-ice" value="true"/>   <!-- also tried false -->
<param name="inbound-late-negotiation" value="true"/>

<param name="ext-sip-ip" value="$${external_sip_ip}"/>
<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>

---

vars.xml

<X-PRE-PROCESS cmd="set" data="domain=10.10.4.104"/>
<X-PRE-PROCESS cmd="set" data="external_sip_ip=10.10.4.104"/>
<X-PRE-PROCESS cmd="set" data="external_rtp_ip=10.10.4.104"/>

---

Client SDP Characteristics

• Uses WebRTC with DTLS-SRTP

• Codecs offered: OPUS, PCMU, PCMA

• rtcp-mux enabled

• Tried both:

  • With STUN servers
  • Without any ICE servers

---

What I Tried

• Correct external IP in vars.xml
• Opened RTP ports 16384–32768 UDP
• Enabled WebRTC params in SIP profile
• Disabled ICE for testing
• Removed STUN from client
• Forced codecs (OPUS, PCMU, PCMA)
• Restarted profile and server multiple times

But media negotiation still fails with INCOMPATIBLE_DESTINATION.

---

Question

What exact SIP profile / media configuration is required in FreeSWITCH 1.10.x to properly accept WebRTC (JsSIP) SDP with DTLS-SRTP?

Is there any missing parameter for:

• DTLS
• SRTP
• ICE
• Candidate ACL
• NAT handling

that could cause this consistent “no suitable candidates” / “incompatible destination” error?

Any guidance or minimal working WebRTC profile example for 1.10.x would be very helpful.

Thanks in advance.

[401U]

have you tried to modify the apply-candidate-acl params in profile config? you can set an ACL list into that param, which include the cidr 10.10.4.0/24