Linphone Android RTP/Voicemail Audio Stutter with FreeSWITCH/FusionPBX

Omid Farahmand via Linphone-users <[email protected]> Fri, 5 Jun 2026 18:12:21 -0400
Newsgroups	gmane.comp.voip.linphone.user
Organization	Netlayers Inc.
Message-ID	<[email protected]>
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Hello Linphone Support Team,

=20

We are testing Linphone for our own internal SIP softphone use with a =
FusionPBX/FreeSWITCH server.

=20

We are experiencing an audio quality issue mainly with voicemail and =
server generated audio. The issue happens both in the official Linphone =
Android app and in our own Android app built using the Linphone SDK.

=20

Using the same SIP account on the same FusionPBX/FreeSWITCH server, =
Zoiper and desk phones play the voicemail audio cleanly, but the =
official Linphone app and our Linphone SDK-based app both experience =
shaky, stuttering, or cutting voicemail audio.

=20

What we observed:

=20

- Server: FusionPBX / FreeSWITCH

- Client: Official Linphone Android app and our Android app using =
Linphone SDK

- Codec during voicemail: PCMU / G.711

- Packet loss in Wireshark: 0%

- RTP timing: uneven packet spacing with large gaps

- Max inter-packet delta observed: around 500 ms

- Expected PCMU packet rate: around 50 packets/sec for 20 ms ptime

- Observed packet rate was much lower during the problematic voicemail =
stream

- Increasing the Linphone jitter buffer up to around 800 ms did not =
fully fix the voicemail stutter

=20

This makes us think the FreeSWITCH server may be producing uneven or =
starved RTP during voicemail playback, but we are trying to understand =
why Linphone exposes the issue more severely than Zoiper or desk phones.

=20

Questions:

=20

1. Does Linphone Android or Linphone SDK have packet-loss concealment or =
audio gap concealment settings that can be tuned for PCMU/G.711?

2. Are there recommended jitter buffer, RTP, or audio stream settings =
for handling bursty or starved RTP from FreeSWITCH?

3. Are there Linphone SDK settings related to adaptive jitter buffer, =
PLC, RTP timing, audio playback buffering, or media stream recovery that =
we should test?

4. What Linphone logs would be most useful for diagnosing RTP/audio =
playback behavior during voicemail?

5. Are there known compatibility considerations between Linphone and =
FreeSWITCH voicemail or server-generated media?

=20

Thank you,

Omid Farahmand=20
TechTeam-NetLayers Inc.
=F0=9F=93=9E +1 (888) 208-8282
=E2=9C=89 [email protected]
=F0=9F=8C=90  <http://www.netlayers.ca/> www.netlayers.ca
212-7163 Yonge St., Thornhill, ON L3T 0C6

=20


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link=3D"#467886" vlink=3D"#96607D" style=3D'word-wrap:break-word'><div =
class=3DWordSection1><p class=3DMsoNormal>Hello Linphone Support =
Team,<o:p></o:p></p><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p =
class=3DMsoNormal>We are testing Linphone for our own internal SIP =
softphone use with a FusionPBX/FreeSWITCH server.<o:p></o:p></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>We are =
experiencing an audio quality issue mainly with voicemail and server =
generated audio. The issue happens both in the official Linphone Android =
app and in our own Android app built using the Linphone =
SDK.<o:p></o:p></p><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p =
class=3DMsoNormal>Using the same SIP account on the same =
FusionPBX/FreeSWITCH server, Zoiper and desk phones play the voicemail =
audio cleanly, but the official Linphone app and our Linphone SDK-based =
app both experience shaky, stuttering, or cutting voicemail =
audio.<o:p></o:p></p><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p =
class=3DMsoNormal>What we observed:<o:p></o:p></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>- Server: =
FusionPBX / FreeSWITCH<o:p></o:p></p><p class=3DMsoNormal>- Client: =
Official Linphone Android app and our Android app using Linphone =
SDK<o:p></o:p></p><p class=3DMsoNormal>- Codec during voicemail: PCMU / =
G.711<o:p></o:p></p><p class=3DMsoNormal>- Packet loss in Wireshark: =
0%<o:p></o:p></p><p class=3DMsoNormal>- RTP timing: uneven packet =
spacing with large gaps<o:p></o:p></p><p class=3DMsoNormal>- Max =
inter-packet delta observed: around 500 ms<o:p></o:p></p><p =
class=3DMsoNormal>- Expected PCMU packet rate: around 50 packets/sec for =
20 ms ptime<o:p></o:p></p><p class=3DMsoNormal>- Observed packet rate =
was much lower during the problematic voicemail stream<o:p></o:p></p><p =
class=3DMsoNormal>- Increasing the Linphone jitter buffer up to around =
800 ms did not fully fix the voicemail stutter<o:p></o:p></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>This makes =
us think the FreeSWITCH server may be producing uneven or starved RTP =
during voicemail playback, but we are trying to understand why Linphone =
exposes the issue more severely than Zoiper or desk =
phones.<o:p></o:p></p><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p =
class=3DMsoNormal>Questions:<o:p></o:p></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>1. Does =
Linphone Android or Linphone SDK have packet-loss concealment or audio =
gap concealment settings that can be tuned for =
PCMU/G.711?<o:p></o:p></p><p class=3DMsoNormal>2. Are there recommended =
jitter buffer, RTP, or audio stream settings for handling bursty or =
starved RTP from FreeSWITCH?<o:p></o:p></p><p class=3DMsoNormal>3. Are =
there Linphone SDK settings related to adaptive jitter buffer, PLC, RTP =
timing, audio playback buffering, or media stream recovery that we =
should test?<o:p></o:p></p><p class=3DMsoNormal>4. What Linphone logs =
would be most useful for diagnosing RTP/audio playback behavior during =
voicemail?<o:p></o:p></p><p class=3DMsoNormal>5. Are there known =
compatibility considerations between Linphone and FreeSWITCH voicemail =
or server-generated media?<o:p></o:p></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>Thank =
you,<o:p></o:p></p><p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><b><span =
style=3D'mso-fareast-language:EN-CA'>Omid Farahmand</span></b><span =
style=3D'mso-fareast-language:EN-CA'> <br></span><i><span =
style=3D'font-size:11.0pt;mso-fareast-language:EN-CA'>TechTeam-NetLayers =
Inc.</span></i><span =
style=3D'font-size:11.0pt;mso-fareast-language:EN-CA'><br></span><span =
style=3D'font-size:11.0pt;font-family:"Segoe UI =
Emoji",sans-serif;mso-fareast-language:EN-CA'>&#128222;</span><span =
style=3D'font-size:11.0pt;mso-fareast-language:EN-CA'> +1 (888) =
208-8282<br></span><span style=3D'font-size:11.0pt;font-family:"Segoe UI =
Emoji",sans-serif;color:black;mso-fareast-language:EN-CA'>=E2=9C=89</span=
><span =
style=3D'font-size:11.0pt;color:black;mso-fareast-language:EN-CA'> =
[email protected]<br></span><span =
style=3D'font-size:11.0pt;font-family:"Segoe UI =
Emoji",sans-serif;color:black;mso-fareast-language:EN-CA'>&#127760;</span=
><span =
style=3D'font-size:11.0pt;color:black;mso-fareast-language:EN-CA'> <a =
href=3D"http://www.netlayers.ca/"><span =
style=3D'color:black;text-decoration:none'>www.netlayers.ca</span></a></s=
pan><span =
style=3D'font-size:11.0pt;mso-fareast-language:EN-CA'><br><i>212-7163 =
Yonge St., Thornhill, ON L3T 0C6</i></span><span =
style=3D'font-size:11.0pt;mso-fareast-language:EN-CA'><o:p></o:p></span><=
/p><p class=3DMsoNormal><o:p>&nbsp;</o:p></p></div></body></html>
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