[jitsi-users] Music on hold issues with stable jitsi as SIP client


#1

Hi!

We have a strange problem with music on hold in our VoIP/SIP setup. When an inbound call is put on hold (either because of a transfer or by locally holing the call by pressing the pause button in jitsis call window), the music is stuttering/stammering in various degrees (i.e. sometimes the music plays well, sometimes it is just an annoying sound mess interrupted by silence. The quality of playback even changes when holding and unholding the same call several times.)

We use the latest stable 32 bit version of Jitsi on Windows 7x64 clients.
The SIP server is Asterisk 13.7.2 inside a VM (asterisk.conf has internal_timing unset, defaults to "yes"). The MOH files are WAV mono 8000 Hz 16bit PCL encoded.

The problem only occurs with Jitsi as a client when the audiosystem is either WASAPI or PortAudio. Using Audiosilence (for testing), the issue is gone. Other softphones (3CXPhone) or OSs (Linux) also work.

It is unlikely, that the asterisk server is (alone) to blame, because other music files play perfectly well (for voicemail and queues). Even the same (working) files when used for MOH are not played back well.

We even tried different codecs (ULAW, ALAW, GSM) without any notable differences.

Any hints or suggestions on this issue are welcome. The callers are really getting angry on the annoying sounds they hear when put on hold.

Kind regards and thanks in advance,

Peter


#2

Hi

Just to make sure I understand this problem correctly:

External (PSTN) caller<->Asterisk<->Jitsi

The external caller hears distorted sound when Jitsi puts the call on hold using the pause button?

Info

Freundliche Grüsse,
Ingo Bauersachs

-- sent from my mobile

···

On 8 Nov 2016, at 16:54, Peter Schmidt <peter.schmidt@bixa.cc> wrote:

Hi!

We have a strange problem with music on hold in our VoIP/SIP setup. When an inbound call is put on hold (either because of a transfer or by locally holing the call by pressing the pause button in jitsis call window), the music is stuttering/stammering in various degrees (i.e. sometimes the music plays well, sometimes it is just an annoying sound mess interrupted by silence. The quality of playback even changes when holding and unholding the same call several times.)

We use the latest stable 32 bit version of Jitsi on Windows 7x64 clients.
The SIP server is Asterisk 13.7.2 inside a VM (asterisk.conf has internal_timing unset, defaults to "yes"). The MOH files are WAV mono 8000 Hz 16bit PCL encoded.

The problem only occurs with Jitsi as a client when the audiosystem is either WASAPI or PortAudio. Using Audiosilence (for testing), the issue is gone. Other softphones (3CXPhone) or OSs (Linux) also work.

It is unlikely, that the asterisk server is (alone) to blame, because other music files play perfectly well (for voicemail and queues). Even the same (working) files when used for MOH are not played back well.

We even tried different codecs (ULAW, ALAW, GSM) without any notable differences.

Any hints or suggestions on this issue are welcome. The callers are really getting angry on the annoying sounds they hear when put on hold.

Kind regards and thanks in advance,

Peter

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users mailing list
users@jitsi.org
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#3

Hi Ingo,

exactly, almost. We don't have a PSTN line but an external SIP provider, its (any medium) caller <-> external SIP provider <-> Asterisk <-> Jitsi (if that makes a difference).

The caller sometimes hears
  aaaaaaaa bbbbbbbb cccccccc (as supposed to be)
and sometimes
  aa-aa-aa-aa bb-bb-bb-bb cc-cc-cc-cc
or even
  -----a------a------a-- [...]

I have just tried the latest nightly version jitsi-2.9.5534-x86 (as an update without prior uninstallation or removal of config/cache files) and the problem persist.

Freundliche Grüße,
Peter

···

Am 2016-11-08 um 16:58 schrieb Ingo Bauersachs:

Hi

Just to make sure I understand this problem correctly:

External (PSTN) caller<->Asterisk<->Jitsi

The external caller hears distorted sound when Jitsi puts the call on hold using the pause button?

Info

Freundliche Grüsse,
Ingo Bauersachs

-- sent from my mobile

On 8 Nov 2016, at 16:54, Peter Schmidt <peter.schmidt@bixa.cc> wrote:

Hi!

We have a strange problem with music on hold in our VoIP/SIP setup. When an inbound call is put on hold (either because of a transfer or by locally holing the call by pressing the pause button in jitsis call window), the music is stuttering/stammering in various degrees (i.e. sometimes the music plays well, sometimes it is just an annoying sound mess interrupted by silence. The quality of playback even changes when holding and unholding the same call several times.)

We use the latest stable 32 bit version of Jitsi on Windows 7x64 clients.
The SIP server is Asterisk 13.7.2 inside a VM (asterisk.conf has internal_timing unset, defaults to "yes"). The MOH files are WAV mono 8000 Hz 16bit PCL encoded.

The problem only occurs with Jitsi as a client when the audiosystem is either WASAPI or PortAudio. Using Audiosilence (for testing), the issue is gone. Other softphones (3CXPhone) or OSs (Linux) also work.

It is unlikely, that the asterisk server is (alone) to blame, because other music files play perfectly well (for voicemail and queues). Even the same (working) files when used for MOH are not played back well.

We even tried different codecs (ULAW, ALAW, GSM) without any notable differences.

Any hints or suggestions on this issue are welcome. The callers are really getting angry on the annoying sounds they hear when put on hold.

Kind regards and thanks in advance,

Peter

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users mailing list
users@jitsi.org
Unsubscribe instructions and other list options:
http://lists.jitsi.org/mailman/listinfo/users

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#4

That was what I meant with external, but thanks for the clarification.

I don't see how Jitsi could cause distorted sound on Asterisks side because the MoH comes from there... anyway, can you provide network traces and Jitsi's logs of an affected call, as well as a set of logs from a clean call (Linux or another phone)?

If you don't want them on the list, send them to me directly.

Ingo

Freundliche Grüsse,
Ingo Bauersachs

-- sent from my mobile

···

On 8 Nov 2016, at 17:15, Peter Schmidt <peter.schmidt@bixa.cc> wrote:

Hi Ingo,

exactly, almost. We don't have a PSTN line but an external SIP provider, its (any medium) caller <-> external SIP provider <-> Asterisk <-> Jitsi (if that makes a difference).

The caller sometimes hears
aaaaaaaa bbbbbbbb cccccccc (as supposed to be)
and sometimes
aa-aa-aa-aa bb-bb-bb-bb cc-cc-cc-cc
or even
-----a------a------a-- [...]

I have just tried the latest nightly version jitsi-2.9.5534-x86 (as an update without prior uninstallation or removal of config/cache files) and the problem persist.

Freundliche Grüße,
Peter

Am 2016-11-08 um 16:58 schrieb Ingo Bauersachs:
Hi

Just to make sure I understand this problem correctly:

External (PSTN) caller<->Asterisk<->Jitsi

The external caller hears distorted sound when Jitsi puts the call on hold using the pause button?

Info

Freundliche Grüsse,
Ingo Bauersachs

-- sent from my mobile

On 8 Nov 2016, at 16:54, Peter Schmidt <peter.schmidt@bixa.cc> wrote:

Hi!

We have a strange problem with music on hold in our VoIP/SIP setup. When an inbound call is put on hold (either because of a transfer or by locally holing the call by pressing the pause button in jitsis call window), the music is stuttering/stammering in various degrees (i.e. sometimes the music plays well, sometimes it is just an annoying sound mess interrupted by silence. The quality of playback even changes when holding and unholding the same call several times.)

We use the latest stable 32 bit version of Jitsi on Windows 7x64 clients.
The SIP server is Asterisk 13.7.2 inside a VM (asterisk.conf has internal_timing unset, defaults to "yes"). The MOH files are WAV mono 8000 Hz 16bit PCL encoded.

The problem only occurs with Jitsi as a client when the audiosystem is either WASAPI or PortAudio. Using Audiosilence (for testing), the issue is gone. Other softphones (3CXPhone) or OSs (Linux) also work.

It is unlikely, that the asterisk server is (alone) to blame, because other music files play perfectly well (for voicemail and queues). Even the same (working) files when used for MOH are not played back well.

We even tried different codecs (ULAW, ALAW, GSM) without any notable differences.

Any hints or suggestions on this issue are welcome. The callers are really getting angry on the annoying sounds they hear when put on hold.

Kind regards and thanks in advance,

Peter

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users mailing list
users@jitsi.org
Unsubscribe instructions and other list options:
http://lists.jitsi.org/mailman/listinfo/users

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users mailing list
users@jitsi.org
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#5

Hi Ingo,

thanks for your help. I now captured the asterisk SIP debug, tcpdump (pcap) on the router, tcpdump (pcap) on the asterisk server, jitsi log and jitsis pcap.

All in 2 scenarios:
1. regular setup with stuttering MOH
2. audiosilence setup with good MOH

Is this what you wanted me to generate?

Do you know a tool for anonymizing pcap files?

Kind regards,
Peter

···

Am 2016-11-09 um 14:34 schrieb Ingo Bauersachs:

That was what I meant with external, but thanks for the clarification.

I don't see how Jitsi could cause distorted sound on Asterisks side because the MoH comes from there... anyway, can you provide network traces and Jitsi's logs of an affected call, as well as a set of logs from a clean call (Linux or another phone)?

If you don't want them on the list, send them to me directly.

Ingo

Freundliche Grüsse,
Ingo Bauersachs

-- sent from my mobile

On 8 Nov 2016, at 17:15, Peter Schmidt <peter.schmidt@bixa.cc> wrote:

Hi Ingo,

exactly, almost. We don't have a PSTN line but an external SIP provider, its (any medium) caller <-> external SIP provider <-> Asterisk <-> Jitsi (if that makes a difference).

The caller sometimes hears
aaaaaaaa bbbbbbbb cccccccc (as supposed to be)
and sometimes
aa-aa-aa-aa bb-bb-bb-bb cc-cc-cc-cc
or even
-----a------a------a-- [...]

I have just tried the latest nightly version jitsi-2.9.5534-x86 (as an update without prior uninstallation or removal of config/cache files) and the problem persist.

Freundliche Grüße,
Peter

Am 2016-11-08 um 16:58 schrieb Ingo Bauersachs:
Hi

Just to make sure I understand this problem correctly:

External (PSTN) caller<->Asterisk<->Jitsi

The external caller hears distorted sound when Jitsi puts the call on hold using the pause button?

Info

Freundliche Grüsse,
Ingo Bauersachs

-- sent from my mobile

On 8 Nov 2016, at 16:54, Peter Schmidt <peter.schmidt@bixa.cc> wrote:

Hi!

We have a strange problem with music on hold in our VoIP/SIP setup. When an inbound call is put on hold (either because of a transfer or by locally holing the call by pressing the pause button in jitsis call window), the music is stuttering/stammering in various degrees (i.e. sometimes the music plays well, sometimes it is just an annoying sound mess interrupted by silence. The quality of playback even changes when holding and unholding the same call several times.)

We use the latest stable 32 bit version of Jitsi on Windows 7x64 clients.
The SIP server is Asterisk 13.7.2 inside a VM (asterisk.conf has internal_timing unset, defaults to "yes"). The MOH files are WAV mono 8000 Hz 16bit PCL encoded.

The problem only occurs with Jitsi as a client when the audiosystem is either WASAPI or PortAudio. Using Audiosilence (for testing), the issue is gone. Other softphones (3CXPhone) or OSs (Linux) also work.

It is unlikely, that the asterisk server is (alone) to blame, because other music files play perfectly well (for voicemail and queues). Even the same (working) files when used for MOH are not played back well.

We even tried different codecs (ULAW, ALAW, GSM) without any notable differences.

Any hints or suggestions on this issue are welcome. The callers are really getting angry on the annoying sounds they hear when put on hold.

Kind regards and thanks in advance,

Peter

_______________________________________________
users mailing list
users@jitsi.org
Unsubscribe instructions and other list options:
http://lists.jitsi.org/mailman/listinfo/users

_______________________________________________
users mailing list
users@jitsi.org
Unsubscribe instructions and other list options:
http://lists.jitsi.org/mailman/listinfo/users

_______________________________________________
users mailing list
users@jitsi.org
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#6

Hi Ingo,

By looking at the tcpdumps on the router (i.e. who talks to the external SIP provider), I can see clearly that in case of the working audiosilence situation, the RTP stream during hold comes only from the asterisk server. No RTP packets from the client are recorded during hold. This is an extremely clean SIP/RTP trace.

But in the normal setup with WASAPI, when the caller is put on hold, the client still transmits RTP data to the SIP provider aswell. Further, the provider talks RTP to the client AND to the server. So we have 2 concurring bidirectional RTP streams sending data to the same port of the provider.

I've just recorded another call on the router and was put on hold several times until I could hear a clean MOH (it took 5 attempts):

#4: the 4th time put on hold was rather extrem:
I can see 2 bidirectional RTP streams.

#5: MOH was working this time:
I can see bidirectional RTP data between client and provider and one-directional RTP data from server to provider (MOH?!)

So there IS a difference. The question is: WHY.

Any suggestions? Do you need any more data?

Kind regards,
Peter

···

Am 2016-11-14 um 17:53 schrieb Peter Schmidt:

Hi Ingo,

thanks for your help. I now captured the asterisk SIP debug, tcpdump
(pcap) on the router, tcpdump (pcap) on the asterisk server, jitsi log
and jitsis pcap.

All in 2 scenarios:
1. regular setup with stuttering MOH
2. audiosilence setup with good MOH

Is this what you wanted me to generate?

Do you know a tool for anonymizing pcap files?

Kind regards,
Peter

Am 2016-11-09 um 14:34 schrieb Ingo Bauersachs:

That was what I meant with external, but thanks for the clarification.

I don't see how Jitsi could cause distorted sound on Asterisks side
because the MoH comes from there... anyway, can you provide network
traces and Jitsi's logs of an affected call, as well as a set of logs
from a clean call (Linux or another phone)?

If you don't want them on the list, send them to me directly.

Ingo

Freundliche Grüsse,
Ingo Bauersachs

-- sent from my mobile

On 8 Nov 2016, at 17:15, Peter Schmidt <peter.schmidt@bixa.cc> wrote:

Hi Ingo,

exactly, almost. We don't have a PSTN line but an external SIP
provider, its (any medium) caller <-> external SIP provider <->
Asterisk <-> Jitsi (if that makes a difference).

The caller sometimes hears
aaaaaaaa bbbbbbbb cccccccc (as supposed to be)
and sometimes
aa-aa-aa-aa bb-bb-bb-bb cc-cc-cc-cc
or even
-----a------a------a-- [...]

I have just tried the latest nightly version jitsi-2.9.5534-x86 (as
an update without prior uninstallation or removal of config/cache
files) and the problem persist.

Freundliche Grüße,
Peter

Am 2016-11-08 um 16:58 schrieb Ingo Bauersachs:
Hi

Just to make sure I understand this problem correctly:

External (PSTN) caller<->Asterisk<->Jitsi

The external caller hears distorted sound when Jitsi puts the call
on hold using the pause button?

Info

Freundliche Grüsse,
Ingo Bauersachs

-- sent from my mobile

On 8 Nov 2016, at 16:54, Peter Schmidt <peter.schmidt@bixa.cc> wrote:

Hi!

We have a strange problem with music on hold in our VoIP/SIP setup.
When an inbound call is put on hold (either because of a transfer
or by locally holing the call by pressing the pause button in
jitsis call window), the music is stuttering/stammering in various
degrees (i.e. sometimes the music plays well, sometimes it is just
an annoying sound mess interrupted by silence. The quality of
playback even changes when holding and unholding the same call
several times.)

We use the latest stable 32 bit version of Jitsi on Windows 7x64
clients.
The SIP server is Asterisk 13.7.2 inside a VM (asterisk.conf has
internal_timing unset, defaults to "yes"). The MOH files are WAV
mono 8000 Hz 16bit PCL encoded.

The problem only occurs with Jitsi as a client when the audiosystem
is either WASAPI or PortAudio. Using Audiosilence (for testing),
the issue is gone. Other softphones (3CXPhone) or OSs (Linux) also
work.

It is unlikely, that the asterisk server is (alone) to blame,
because other music files play perfectly well (for voicemail and
queues). Even the same (working) files when used for MOH are not
played back well.

We even tried different codecs (ULAW, ALAW, GSM) without any
notable differences.

Any hints or suggestions on this issue are welcome. The callers are
really getting angry on the annoying sounds they hear when put on
hold.

Kind regards and thanks in advance,

Peter

_______________________________________________
users mailing list
users@jitsi.org
Unsubscribe instructions and other list options:
http://lists.jitsi.org/mailman/listinfo/users

_______________________________________________
users mailing list
users@jitsi.org
Unsubscribe instructions and other list options:
http://lists.jitsi.org/mailman/listinfo/users

_______________________________________________
users mailing list
users@jitsi.org
Unsubscribe instructions and other list options:
http://lists.jitsi.org/mailman/listinfo/users

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users mailing list
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#7

Do you use Asterisk's directmedia option? If so, that doesn't work with MoH.

Ingo

Freundliche Grüsse,
Ingo Bauersachs

-- sent from my mobile

···

On 15 Nov 2016, at 12:16, Peter Schmidt <peter.schmidt@bixa.cc> wrote:

Hi Ingo,

By looking at the tcpdumps on the router (i.e. who talks to the external SIP provider), I can see clearly that in case of the working audiosilence situation, the RTP stream during hold comes only from the asterisk server. No RTP packets from the client are recorded during hold. This is an extremely clean SIP/RTP trace.

But in the normal setup with WASAPI, when the caller is put on hold, the client still transmits RTP data to the SIP provider aswell. Further, the provider talks RTP to the client AND to the server. So we have 2 concurring bidirectional RTP streams sending data to the same port of the provider.

I've just recorded another call on the router and was put on hold several times until I could hear a clean MOH (it took 5 attempts):

#4: the 4th time put on hold was rather extrem:
I can see 2 bidirectional RTP streams.

#5: MOH was working this time:
I can see bidirectional RTP data between client and provider and one-directional RTP data from server to provider (MOH?!)

So there IS a difference. The question is: WHY.

Any suggestions? Do you need any more data?

Kind regards,
Peter

Am 2016-11-14 um 17:53 schrieb Peter Schmidt:
Hi Ingo,

thanks for your help. I now captured the asterisk SIP debug, tcpdump
(pcap) on the router, tcpdump (pcap) on the asterisk server, jitsi log
and jitsis pcap.

All in 2 scenarios:
1. regular setup with stuttering MOH
2. audiosilence setup with good MOH

Is this what you wanted me to generate?

Do you know a tool for anonymizing pcap files?

Kind regards,
Peter

Am 2016-11-09 um 14:34 schrieb Ingo Bauersachs:
That was what I meant with external, but thanks for the clarification.

I don't see how Jitsi could cause distorted sound on Asterisks side
because the MoH comes from there... anyway, can you provide network
traces and Jitsi's logs of an affected call, as well as a set of logs
from a clean call (Linux or another phone)?

If you don't want them on the list, send them to me directly.

Ingo

Freundliche Grüsse,
Ingo Bauersachs

-- sent from my mobile

On 8 Nov 2016, at 17:15, Peter Schmidt <peter.schmidt@bixa.cc> wrote:

Hi Ingo,

exactly, almost. We don't have a PSTN line but an external SIP
provider, its (any medium) caller <-> external SIP provider <->
Asterisk <-> Jitsi (if that makes a difference).

The caller sometimes hears
aaaaaaaa bbbbbbbb cccccccc (as supposed to be)
and sometimes
aa-aa-aa-aa bb-bb-bb-bb cc-cc-cc-cc
or even
-----a------a------a-- [...]

I have just tried the latest nightly version jitsi-2.9.5534-x86 (as
an update without prior uninstallation or removal of config/cache
files) and the problem persist.

Freundliche Grüße,
Peter

Am 2016-11-08 um 16:58 schrieb Ingo Bauersachs:
Hi

Just to make sure I understand this problem correctly:

External (PSTN) caller<->Asterisk<->Jitsi

The external caller hears distorted sound when Jitsi puts the call
on hold using the pause button?

Info

Freundliche Grüsse,
Ingo Bauersachs

-- sent from my mobile

On 8 Nov 2016, at 16:54, Peter Schmidt <peter.schmidt@bixa.cc> wrote:

Hi!

We have a strange problem with music on hold in our VoIP/SIP setup.
When an inbound call is put on hold (either because of a transfer
or by locally holing the call by pressing the pause button in
jitsis call window), the music is stuttering/stammering in various
degrees (i.e. sometimes the music plays well, sometimes it is just
an annoying sound mess interrupted by silence. The quality of
playback even changes when holding and unholding the same call
several times.)

We use the latest stable 32 bit version of Jitsi on Windows 7x64
clients.
The SIP server is Asterisk 13.7.2 inside a VM (asterisk.conf has
internal_timing unset, defaults to "yes"). The MOH files are WAV
mono 8000 Hz 16bit PCL encoded.

The problem only occurs with Jitsi as a client when the audiosystem
is either WASAPI or PortAudio. Using Audiosilence (for testing),
the issue is gone. Other softphones (3CXPhone) or OSs (Linux) also
work.

It is unlikely, that the asterisk server is (alone) to blame,
because other music files play perfectly well (for voicemail and
queues). Even the same (working) files when used for MOH are not
played back well.

We even tried different codecs (ULAW, ALAW, GSM) without any
notable differences.

Any hints or suggestions on this issue are welcome. The callers are
really getting angry on the annoying sounds they hear when put on
hold.

Kind regards and thanks in advance,

Peter

_______________________________________________
users mailing list
users@jitsi.org
Unsubscribe instructions and other list options:
http://lists.jitsi.org/mailman/listinfo/users

_______________________________________________
users mailing list
users@jitsi.org
Unsubscribe instructions and other list options:
http://lists.jitsi.org/mailman/listinfo/users

_______________________________________________
users mailing list
users@jitsi.org
Unsubscribe instructions and other list options:
http://lists.jitsi.org/mailman/listinfo/users

_______________________________________________
users mailing list
users@jitsi.org
Unsubscribe instructions and other list options:
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#8

Hi Ingo,

directmedia is not set in sip.conf, which means it defaults to YES in my understanding of the sample sip.conf.

I also see the SIP INVITE messages when the call is put on hold and back again.

But I put 'directmedia = no' in the global section of sip.conf and did '/etc/init.d/asterisk reload' but nothing has changed.

Googleing for the topic is also hard as nobody really explains where and why to put this setting in the sip.conf file.
Do you have any further suggestions?

Kind regards and thanks a lot.

Peter

···

Am 2016-11-15 um 13:05 schrieb Ingo Bauersachs:

Do you use Asterisk's directmedia option? If so, that doesn't work with MoH.

Ingo

Freundliche Grüsse,
Ingo Bauersachs

-- sent from my mobile

On 15 Nov 2016, at 12:16, Peter Schmidt <peter.schmidt@bixa.cc> wrote:

Hi Ingo,

By looking at the tcpdumps on the router (i.e. who talks to the external SIP provider), I can see clearly that in case of the working audiosilence situation, the RTP stream during hold comes only from the asterisk server. No RTP packets from the client are recorded during hold. This is an extremely clean SIP/RTP trace.

But in the normal setup with WASAPI, when the caller is put on hold, the client still transmits RTP data to the SIP provider aswell. Further, the provider talks RTP to the client AND to the server. So we have 2 concurring bidirectional RTP streams sending data to the same port of the provider.

I've just recorded another call on the router and was put on hold several times until I could hear a clean MOH (it took 5 attempts):

#4: the 4th time put on hold was rather extrem:
I can see 2 bidirectional RTP streams.

#5: MOH was working this time:
I can see bidirectional RTP data between client and provider and one-directional RTP data from server to provider (MOH?!)

So there IS a difference. The question is: WHY.

Any suggestions? Do you need any more data?

Kind regards,
Peter

Am 2016-11-14 um 17:53 schrieb Peter Schmidt:
Hi Ingo,

thanks for your help. I now captured the asterisk SIP debug, tcpdump
(pcap) on the router, tcpdump (pcap) on the asterisk server, jitsi log
and jitsis pcap.

All in 2 scenarios:
1. regular setup with stuttering MOH
2. audiosilence setup with good MOH

Is this what you wanted me to generate?

Do you know a tool for anonymizing pcap files?

Kind regards,
Peter

Am 2016-11-09 um 14:34 schrieb Ingo Bauersachs:
That was what I meant with external, but thanks for the clarification.

I don't see how Jitsi could cause distorted sound on Asterisks side
because the MoH comes from there... anyway, can you provide network
traces and Jitsi's logs of an affected call, as well as a set of logs
from a clean call (Linux or another phone)?

If you don't want them on the list, send them to me directly.

Ingo

Freundliche Grüsse,
Ingo Bauersachs

-- sent from my mobile

On 8 Nov 2016, at 17:15, Peter Schmidt <peter.schmidt@bixa.cc> wrote:

Hi Ingo,

exactly, almost. We don't have a PSTN line but an external SIP
provider, its (any medium) caller <-> external SIP provider <->
Asterisk <-> Jitsi (if that makes a difference).

The caller sometimes hears
aaaaaaaa bbbbbbbb cccccccc (as supposed to be)
and sometimes
aa-aa-aa-aa bb-bb-bb-bb cc-cc-cc-cc
or even
-----a------a------a-- [...]

I have just tried the latest nightly version jitsi-2.9.5534-x86 (as
an update without prior uninstallation or removal of config/cache
files) and the problem persist.

Freundliche Grüße,
Peter

Am 2016-11-08 um 16:58 schrieb Ingo Bauersachs:
Hi

Just to make sure I understand this problem correctly:

External (PSTN) caller<->Asterisk<->Jitsi

The external caller hears distorted sound when Jitsi puts the call
on hold using the pause button?

Info

Freundliche Grüsse,
Ingo Bauersachs

-- sent from my mobile

On 8 Nov 2016, at 16:54, Peter Schmidt <peter.schmidt@bixa.cc> wrote:

Hi!

We have a strange problem with music on hold in our VoIP/SIP setup.
When an inbound call is put on hold (either because of a transfer
or by locally holing the call by pressing the pause button in
jitsis call window), the music is stuttering/stammering in various
degrees (i.e. sometimes the music plays well, sometimes it is just
an annoying sound mess interrupted by silence. The quality of
playback even changes when holding and unholding the same call
several times.)

We use the latest stable 32 bit version of Jitsi on Windows 7x64
clients.
The SIP server is Asterisk 13.7.2 inside a VM (asterisk.conf has
internal_timing unset, defaults to "yes"). The MOH files are WAV
mono 8000 Hz 16bit PCL encoded.

The problem only occurs with Jitsi as a client when the audiosystem
is either WASAPI or PortAudio. Using Audiosilence (for testing),
the issue is gone. Other softphones (3CXPhone) or OSs (Linux) also
work.

It is unlikely, that the asterisk server is (alone) to blame,
because other music files play perfectly well (for voicemail and
queues). Even the same (working) files when used for MOH are not
played back well.

We even tried different codecs (ULAW, ALAW, GSM) without any
notable differences.

Any hints or suggestions on this issue are welcome. The callers are
really getting angry on the annoying sounds they hear when put on
hold.

Kind regards and thanks in advance,

Peter

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#9

Hi Ingo,

sorry, but I have news:
Settings 'directmedia=no' in [general] context results in all sip peers accepting this setting.

i.e.:
sip show peer <internal user>
...
   DirectMedia : No
...
sip show peer <external sip trunk>
...
   DirectMedia : No
...

Still, MOH is buggy on inbound calls.

Do you have any more ideas?

Kind regards,
Peter

···

Am 2016-11-15 um 14:56 schrieb Peter Schmidt:

Hi Ingo,

directmedia is not set in sip.conf, which means it defaults to YES in my
understanding of the sample sip.conf.

I also see the SIP INVITE messages when the call is put on hold and back
again.

But I put 'directmedia = no' in the global section of sip.conf and did
'/etc/init.d/asterisk reload' but nothing has changed.

Googleing for the topic is also hard as nobody really explains where and
why to put this setting in the sip.conf file.
Do you have any further suggestions?

Kind regards and thanks a lot.

Peter

Am 2016-11-15 um 13:05 schrieb Ingo Bauersachs:

Do you use Asterisk's directmedia option? If so, that doesn't work
with MoH.

Ingo

Freundliche Grüsse,
Ingo Bauersachs

-- sent from my mobile

On 15 Nov 2016, at 12:16, Peter Schmidt <peter.schmidt@bixa.cc> wrote:

Hi Ingo,

By looking at the tcpdumps on the router (i.e. who talks to the
external SIP provider), I can see clearly that in case of the working
audiosilence situation, the RTP stream during hold comes only from
the asterisk server. No RTP packets from the client are recorded
during hold. This is an extremely clean SIP/RTP trace.

But in the normal setup with WASAPI, when the caller is put on hold,
the client still transmits RTP data to the SIP provider aswell.
Further, the provider talks RTP to the client AND to the server. So
we have 2 concurring bidirectional RTP streams sending data to the
same port of the provider.

I've just recorded another call on the router and was put on hold
several times until I could hear a clean MOH (it took 5 attempts):

#4: the 4th time put on hold was rather extrem:
I can see 2 bidirectional RTP streams.

#5: MOH was working this time:
I can see bidirectional RTP data between client and provider and
one-directional RTP data from server to provider (MOH?!)

So there IS a difference. The question is: WHY.

Any suggestions? Do you need any more data?

Kind regards,
Peter

Am 2016-11-14 um 17:53 schrieb Peter Schmidt:
Hi Ingo,

thanks for your help. I now captured the asterisk SIP debug, tcpdump
(pcap) on the router, tcpdump (pcap) on the asterisk server, jitsi log
and jitsis pcap.

All in 2 scenarios:
1. regular setup with stuttering MOH
2. audiosilence setup with good MOH

Is this what you wanted me to generate?

Do you know a tool for anonymizing pcap files?

Kind regards,
Peter

Am 2016-11-09 um 14:34 schrieb Ingo Bauersachs:
That was what I meant with external, but thanks for the clarification.

I don't see how Jitsi could cause distorted sound on Asterisks side
because the MoH comes from there... anyway, can you provide network
traces and Jitsi's logs of an affected call, as well as a set of logs
from a clean call (Linux or another phone)?

If you don't want them on the list, send them to me directly.

Ingo

Freundliche Grüsse,
Ingo Bauersachs

-- sent from my mobile

On 8 Nov 2016, at 17:15, Peter Schmidt <peter.schmidt@bixa.cc> wrote:

Hi Ingo,

exactly, almost. We don't have a PSTN line but an external SIP
provider, its (any medium) caller <-> external SIP provider <->
Asterisk <-> Jitsi (if that makes a difference).

The caller sometimes hears
aaaaaaaa bbbbbbbb cccccccc (as supposed to be)
and sometimes
aa-aa-aa-aa bb-bb-bb-bb cc-cc-cc-cc
or even
-----a------a------a-- [...]

I have just tried the latest nightly version jitsi-2.9.5534-x86 (as
an update without prior uninstallation or removal of config/cache
files) and the problem persist.

Freundliche Grüße,
Peter

Am 2016-11-08 um 16:58 schrieb Ingo Bauersachs:
Hi

Just to make sure I understand this problem correctly:

External (PSTN) caller<->Asterisk<->Jitsi

The external caller hears distorted sound when Jitsi puts the call
on hold using the pause button?

Info

Freundliche Grüsse,
Ingo Bauersachs

-- sent from my mobile

On 8 Nov 2016, at 16:54, Peter Schmidt <peter.schmidt@bixa.cc> >>>>>>>> wrote:

Hi!

We have a strange problem with music on hold in our VoIP/SIP setup.
When an inbound call is put on hold (either because of a transfer
or by locally holing the call by pressing the pause button in
jitsis call window), the music is stuttering/stammering in various
degrees (i.e. sometimes the music plays well, sometimes it is just
an annoying sound mess interrupted by silence. The quality of
playback even changes when holding and unholding the same call
several times.)

We use the latest stable 32 bit version of Jitsi on Windows 7x64
clients.
The SIP server is Asterisk 13.7.2 inside a VM (asterisk.conf has
internal_timing unset, defaults to "yes"). The MOH files are WAV
mono 8000 Hz 16bit PCL encoded.

The problem only occurs with Jitsi as a client when the audiosystem
is either WASAPI or PortAudio. Using Audiosilence (for testing),
the issue is gone. Other softphones (3CXPhone) or OSs (Linux) also
work.

It is unlikely, that the asterisk server is (alone) to blame,
because other music files play perfectly well (for voicemail and
queues). Even the same (working) files when used for MOH are not
played back well.

We even tried different codecs (ULAW, ALAW, GSM) without any
notable differences.

Any hints or suggestions on this issue are welcome. The callers are
really getting angry on the annoying sounds they hear when put on
hold.

Kind regards and thanks in advance,

Peter

_______________________________________________
users mailing list
users@jitsi.org
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users@jitsi.org
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users@jitsi.org
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#10

sorry, but I have news:
Settings 'directmedia=no' in [general] context results in all sip peers
accepting this setting.

i.e.:
sip show peer <internal user>
...
   DirectMedia : No
...
sip show peer <external sip trunk>
...
   DirectMedia : No
...

Still, MOH is buggy on inbound calls.

Do you have any more ideas?

Not really. Can you make these .pcap traces you captured available?

I'm not really a SIP expert, but from what I remember: if Jitsi puts a caller on hold, it sends a reinvite to Asterisk which changes the media directions (sendrecv). Afterwards Asterisk starts playing MoH and anything from the client should be completely irrelevant, the channels shouldn't even be bridged anymore.

Kind regards,
Peter

Ingo