No audio with Linphone v6 (works wi th v5) – RTP sent but not received (no NAT on PBX s ide)
GASEL SARL via Linphone-users <[email protected]> Thu, 9 Apr 2026 08:38:57 +0200
| Newsgroups | gmane.comp.voip.linphone.user |
|---|---|
| Message-ID | <[email protected]> |
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Hi, everybody
I am experiencing a no-audio issue with Linphone v6, while the exact
same configuration works perfectly with Linphone v5 and Zoiper.
*Environment:*
*
Asterisk / FreePBX version: Asterisk 22.8.2
*
Transport: SIP over UDP (PJSIP)
*
Network:
o
PBX is *NOT behind NAT (public IP directly)*
o
Client is connected via *WireGuard VPN*
*
Softphones tested:
o
Zoiper → OK (2-way audio)
o
Linphone v5 → OK
o
Linphone v6 → NO AUDIO
*Asterisk endpoint configuration (relevant parts):*
*
|direct_media = false|
*
|rtp_symmetric = true|
*
|rewrite_contact = true|
*
|force_rport = true|
*
Codec allowed: |alaw| only
*
|rtp_keepalive = 1|
*Observed behavior:*
*
Registration: OK
*
Authentication: OK
*
Calls are established successfully
*
Phone rings and call is answered
*
RTP debug shows continuous audio packets
Example RTP debug:
|Got RTP packet from 79.x.x.x:10174 (type 08) Sent RTP packet to
10.0.88.162:38431 (type 08) |
So:
*
Asterisk receives RTP from the provider
*
Asterisk sends RTP to the Linphone client
*Important detail:*
*
PBX is on a *public IP (no NAT involved on server side)*
*
Only the client is behind VPN/NAT
*Problem:*
*
Linphone v6 produces *no audio at all*
*
No incoming audio heard
*
Remote side also does not receive audio
*Contact / RTP mismatch:*
*
SIP Contact:
|sip:[email protected]:55423 |
*
But RTP is sent to:
|10.0.88.162:38431 |
This suggests Linphone v6 dynamically changes RTP ports and may not
correctly handle symmetric RTP.
*What was tested:*
*
Same account on Linphone v5 → works immediately
*
Same account on Zoiper → works immediately
*
Codec forced to A-Law only → no change
*
VPN connectivity verified (bi-directional OK)
*
No NAT on PBX side → eliminates server-side NAT issues
*Suspected causes:*
*
Linphone v6 RTP handling regression vs v5
*
Incorrect symmetric RTP behavior
*
RTP port mismatch or ignored incoming stream
*
Missing advanced RTP/NAT settings in v6 UI
*Additional notes:*
*
Linphone v6 no longer exposes:
o
RTP symmetric option
o
RTP port configuration
*
This makes it difficult to align behavior with PBX expectations
*Question:*
Has anyone encountered RTP/audio issues with Linphone v6 in a VPN setup
where the PBX is NOT behind NAT?
Is there a way to:
*
Force symmetric RTP?
*
Control RTP port usage?
*
Enable deeper RTP diagnostics?
------------------------------------------------------------------------
*Conclusion:*
The PBX is correctly configured and validated with other clients. The
issue appears specific to Linphone v6 RTP handling, possibly a
regression from v5.
Any insight would be appreciated.
Many thanks !
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<p>Hi, everybody</p>
<p>I am experiencing a no-audio issue with Linphone v6, while the
exact same configuration works perfectly with Linphone v5 and
Zoiper.</p>
<p><strong>Environment:</strong></p>
<ul>
<li>
<p>Asterisk / FreePBX version: Asterisk 22.8.2</p>
</li>
<li>
<p>Transport: SIP over UDP (PJSIP)</p>
</li>
<li>
<p>Network:</p>
<ul>
<li>
<p>PBX is <strong>NOT behind NAT (public IP directly)</strong></p>
</li>
<li>
<p>Client is connected via <strong>WireGuard VPN</strong></p>
</li>
</ul>
</li>
<li>
<p>Softphones tested:</p>
<ul>
<li>
<p>Zoiper → OK (2-way audio)</p>
</li>
<li>
<p>Linphone v5 → OK</p>
</li>
<li>
<p>Linphone v6 → NO AUDIO</p>
</li>
</ul>
</li>
</ul>
<p><strong>Asterisk endpoint configuration (relevant parts):</strong></p>
<ul>
<li>
<p><code>direct_media = false</code></p>
</li>
<li>
<p><code>rtp_symmetric = true</code></p>
</li>
<li>
<p><code>rewrite_contact = true</code></p>
</li>
<li>
<p><code>force_rport = true</code></p>
</li>
<li>
<p>Codec allowed: <code>alaw</code> only</p>
</li>
<li>
<p><code>rtp_keepalive = 1</code></p>
</li>
</ul>
<p><strong>Observed behavior:</strong></p>
<ul>
<li>
<p>Registration: OK</p>
</li>
<li>
<p>Authentication: OK</p>
</li>
<li>
<p>Calls are established successfully</p>
</li>
<li>
<p>Phone rings and call is answered</p>
</li>
<li>
<p>RTP debug shows continuous audio packets</p>
</li>
</ul>
<p>Example RTP debug:</p>
<pre><code>Got RTP packet from 79.x.x.x:10174 (type 08)
Sent RTP packet to 10.0.88.162:38431 (type 08)
</code></pre>
<p>So:</p>
<ul>
<li>
<p>Asterisk receives RTP from the provider</p>
</li>
<li>
<p>Asterisk sends RTP to the Linphone client</p>
</li>
</ul>
<p><strong>Important detail:</strong></p>
<ul>
<li>
<p>PBX is on a <strong>public IP (no NAT involved on server
side)</strong></p>
</li>
<li>
<p>Only the client is behind VPN/NAT</p>
</li>
</ul>
<p><strong>Problem:</strong></p>
<ul>
<li>
<p>Linphone v6 produces <strong>no audio at all</strong></p>
</li>
<li>
<p>No incoming audio heard</p>
</li>
<li>
<p>Remote side also does not receive audio</p>
</li>
</ul>
<p><strong>Contact / RTP mismatch:</strong></p>
<ul>
<li>
<p>SIP Contact:</p>
<pre><code><a class="moz-txt-link-freetext" href="sip:[email protected]:55423">sip:[email protected]:55423</a>
</code></pre>
</li>
<li>
<p>But RTP is sent to:</p>
<pre><code>10.0.88.162:38431
</code></pre>
</li>
</ul>
<p>This suggests Linphone v6 dynamically changes RTP ports and may
not correctly handle symmetric RTP.</p>
<p><strong>What was tested:</strong></p>
<ul>
<li>
<p>Same account on Linphone v5 → works immediately</p>
</li>
<li>
<p>Same account on Zoiper → works immediately</p>
</li>
<li>
<p>Codec forced to A-Law only → no change</p>
</li>
<li>
<p>VPN connectivity verified (bi-directional OK)</p>
</li>
<li>
<p>No NAT on PBX side → eliminates server-side NAT issues</p>
</li>
</ul>
<p><strong>Suspected causes:</strong></p>
<ul>
<li>
<p>Linphone v6 RTP handling regression vs v5</p>
</li>
<li>
<p>Incorrect symmetric RTP behavior</p>
</li>
<li>
<p>RTP port mismatch or ignored incoming stream</p>
</li>
<li>
<p>Missing advanced RTP/NAT settings in v6 UI</p>
</li>
</ul>
<p><strong>Additional notes:</strong></p>
<ul>
<li>
<p>Linphone v6 no longer exposes:</p>
<ul>
<li>
<p>RTP symmetric option</p>
</li>
<li>
<p>RTP port configuration</p>
</li>
</ul>
</li>
<li>
<p>This makes it difficult to align behavior with PBX
expectations</p>
</li>
</ul>
<p><strong>Question:</strong><br>
Has anyone encountered RTP/audio issues with Linphone v6 in a VPN
setup where the PBX is NOT behind NAT?</p>
<p>Is there a way to:</p>
<ul>
<li>
<p>Force symmetric RTP?</p>
</li>
<li>
<p>Control RTP port usage?</p>
</li>
<li>
<p>Enable deeper RTP diagnostics?</p>
</li>
</ul>
<hr>
<p><strong>Conclusion:</strong><br>
The PBX is correctly configured and validated with other
clients. The issue appears specific to Linphone v6 RTP handling,
possibly a regression from v5.</p>
<p>Any insight would be appreciated.<br>
<br>
Many thanks !<br>
<br>
</p>
</body>
</html>
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