Re: [asterisk-users] Voice broken during calls (again...)

[Michael Maier]

On 23.06.20 at 21:10 Luca Bertoncello wrote:
> Am 23.06.2020 um 21:08 schrieb Michael Maier:
>> On 23.06.20 at 08:05 Luca Bertoncello wrote:
>>> Am 23.06.2020 07:27, schrieb Luca Bertoncello:
>>>
>>> I again
>>>
> Do not change MTU. Probably there will be another problem. I expect
> packet size 1466 would pass and higher will have the same result. It
>>
>> RTP-VoIP-packets never reach this size. Size is about 214 bytes.
> 
> OK, so it must be something other...
> 
> But I really don't have any idea what... :(

Your basic architecture looks good to me - now you have to start the analysis 
of the problem with pcapsipdump and wireshark as I wrote before to get an idea 
what actually happens at
all. You most probably won't come any further without doing any analyzing. You 
have to learn it. It will take some, or even more, time. You can't do it in 
just few hours or maybe
even days or weeks. It is work or even hard work to learn and to do it.

That's my problem: It's impossible for me to assist because I can't see any 
effort on your side to learn. I won't fix your problem. You have to fix it 
yourself. All I can do, is, to
show you a way to *find* your problem (I can't know your problem) and may be to 
give some hints how to fix it (once you've found it). Finding / localizing 
problems and fixing them
are two completely different things. But before you fix a problem, you have to 
know the problem. Therefore: go and find your problem by starting the analysis. 
That's the first thing
to do.


Regards
Michael

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Re: [asterisk-users] Voice broken during calls (again...)

[Luca Bertoncello]

Am 23.06.2020 um 21:08 schrieb Michael Maier:
> On 23.06.20 at 08:05 Luca Bertoncello wrote:
>> Am 23.06.2020 07:27, schrieb Luca Bertoncello:
>>
>> I again
>>
 Do not change MTU. Probably there will be another problem. I expect
 packet size 1466 would pass and higher will have the same result. It
> 
> RTP-VoIP-packets never reach this size. Size is about 214 bytes.

OK, so it must be something other...

But I really don't have any idea what... :(

Thanks
Luca

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Re: [asterisk-users] Voice broken during calls (again...)

[Michael Maier]

On 23.06.20 at 08:05 Luca Bertoncello wrote:
> Am 23.06.2020 07:27, schrieb Luca Bertoncello:
> 
> I again
> 
>>> Do not change MTU. Probably there will be another problem. I expect
>>> packet size 1466 would pass and higher will have the same result. It

RTP-VoIP-packets never reach this size. Size is about 214 bytes.


Regards
Michael

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Re: [asterisk-users] Controlling Asterisk from within the dialplan

[Jöran Vinzens]

Maybe using a fastAGI running as a sidecar. It may have access to systemd
and therefore be able reload or restart the asterisk. As well as take
further action in case something goes wrong. It can be triggered by a call
same way as App system.

BR
jöran

Doug Lytle  schrieb am Di., 23. Juni 2020, 18:19:

> >>> other than using the System() command?
>
> Not that I am aware of,
>
> Doug
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Re: [asterisk-users] ODBC connection failure - can it be fatal?

[Doug Lytle]

>>> Is there any way I can tell Asterisk that an ODBC connection problem is a 
>>> fatal error

Your be best bet would be to do that check in the script that starts up 
Asterisk and maybe a CRON job that periodically tests connectivity.

Doug

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Re: [asterisk-users] Controlling Asterisk from within the dialplan

[Doug Lytle]

>>> other than using the System() command?

Not that I am aware of,

Doug

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[asterisk-users] Controlling Asterisk from within the dialplan

[Antony Stone]

Hi.

Is there any better way of controlling Asterisk itself (by which I mean, 
shutting it down, or telling it to restart) from within the dialplan, other 
than using the System() command?


Regards,


Antony.

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Re: [asterisk-users] Voice broken during calls (again...)

[Luca Bertoncello]

Am 23.06.2020 um 17:04 schrieb Marek Greško:
> I interchanged LAN and LTE in the sentence.

OK...

> Do you have some kind of NAT in fron of asterisk? Or is your asterisk

No, Asterisk has a public IP. No NAT in front of Asterisk...

> having public IP? Could you share sip.conf (without passwords)? One
> LAN client, one LTE and general section.

Of course:

my outgoing configuration:
[pbxluca]
type=peer
defaultuser=
secret=
dtmfmode=rfc2833
host=tel.t-online.de
context=luca_incoming
outboundproxy=tel.t-online.de
port=5060
fromuser=035
fromdomain=tel.t-online.de
usereqphone=yes
canreinvite=yes
insecure=port,invite
nat=no
qualify=yes
qualifyfreq=600
disallow=all
allow=alaw
allow=ulaw

my phone configuration:
; Lucas Telefon
[004935]
fullname = 004935
secret = 
hassip = yes
hasiax = no
hash323 = no
hasmanager = no
callwaiting = no
context = default
host = dynamic
dtmfmode=rfc2833
canreinvite=no
sendrpid=pai
type=friend
nat=force_rport,comedia
qualify=yes
qualifyfreq=60
avpf=no
force_avp=no
icesupport=no
encryption=no
callgroup=1
pickupgroup=1
deny=0.0.0.0/0.0.0.0
permit=192.168.200.0/255.255.255.0
dial=SIP/004935

my mobile phone:
; Lucas Handy
[0049177222]
fullname = 0049177222
secret = 
hassip = yes
hasiax = no
hash323 = no
hasmanager = no
callwaiting = no
context = default
host = dynamic
dtmfmode=rfc2833
canreinvite=no
sendrpid=pai
type=friend
nat=force_rport,comedia
qualify=yes
qualifyfreq=60
avpf=no
force_avp=no
icesupport=no
encryption=no
callgroup=1
pickupgroup=1
dial=SIP/0049177222
allow = all

sip.conf:
[general]
context=default
allowoverlap=no
udpbindaddr=0.0.0.0:25572
tcpenable=yes
tcpbindaddr=0.0.0.0:25572
tlsenable=no
tlsbindaddr=0.0.0.0:25573
transport=udp
srvlookup=no
minexpiry=480
defaultexpiry=480
disallow=all
allow=alaw
allow=ulaw
allow=ilbc
allow=g729
allow=g723
allow=gsm
language=de
alwaysauthreject = yes
tlscertfile=/etc/asterisk/ssl/asterisk.pem
tlscafile=/etc/asterisk/ssl/ca.crt
tlscipher=ALL
tlsclientmethod=tlsv1
callcounter = yes
t38pt_udptl = yes
faxdetect = no
register =>
035::-0...@t-online.de@pbxluca/004935
register =>
0351112::-0...@t-online.de@pbxfax/0049351112
register =>
0351113::-0...@t-online.de@pbxanika/0049351113
register => 5:@messagenet/5
register => lucabertoncello:x@rebvoice/lucabertoncello
jbenable = no
jbmaxsize = 200
jbresyncthreshold = 1000
jbimpl = fixed

Thanks
Luca Bertoncello
(lucab...@lucabert.de)

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Re: [asterisk-users] Voice broken during calls (again...)

[Marek Greško]

I interchanged LAN and LTE in the sentence.

Do you have some kind of NAT in fron of asterisk? Or is your asterisk
having public IP? Could you share sip.conf (without passwords)? One
LAN client, one LTE and general section.

Marek


2020-06-23 16:29 GMT+02:00, Luca Bertoncello :
> Am 23.06.2020 16:22, schrieb Marek Greško:
>> It seems your problems lie in something other. Most probably it is not
>> mtu problem. All my suspections are contradicted. If it is true you
>> have inter vlan voice quality problems, it is definitely something
>> different. Formerly I assumed you were trying only LTE vs LAN using
>> internet.
>
> I'm not sure what you mean with the last sentence...
> I tried to connect to my Asterisk via LAN or via DSL (either via LTE or
> other DSL).
> Then I noticed that if I call another peer in same network (= both peers
> via DSL or both peers in the same VLAN), the quality is very good,
> otherwise is very poor.
>
> But why should Asterisk have problem if the peers are in different
> networks it's for me a really big mistery...
>
> This evening I'll try to capture the pakets in a call between two peers
> connected to Asterisk via LTE, two peers connected in the same LAN and a
> peer connected via LTE and the other in LAN, then maybe it's possible to
> find the problem...
>
> But if you have any other idea, I'm very happy to hear it! ;)
>
> Thanks
> Luca Bertoncello
> (lucab...@lucabert.de)
>
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Re: [asterisk-users] Voice broken during calls (again...)

[Luca Bertoncello]

Am 23.06.2020 16:22, schrieb Marek Greško:

It seems your problems lie in something other. Most probably it is not
mtu problem. All my suspections are contradicted. If it is true you
have inter vlan voice quality problems, it is definitely something
different. Formerly I assumed you were trying only LTE vs LAN using
internet.


I'm not sure what you mean with the last sentence...
I tried to connect to my Asterisk via LAN or via DSL (either via LTE or 
other DSL).
Then I noticed that if I call another peer in same network (= both peers 
via DSL or both peers in the same VLAN), the quality is very good, 
otherwise is very poor.


But why should Asterisk have problem if the peers are in different 
networks it's for me a really big mistery...


This evening I'll try to capture the pakets in a call between two peers 
connected to Asterisk via LTE, two peers connected in the same LAN and a 
peer connected via LTE and the other in LAN, then maybe it's possible to 
find the problem...


But if you have any other idea, I'm very happy to hear it! ;)

Thanks
Luca Bertoncello
(lucab...@lucabert.de)

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Re: [asterisk-users] Voice broken during calls (again...)

[Marek Greško]

It seems your problems lie in something other. Most probably it is not
mtu problem. All my suspections are contradicted. If it is true you
have inter vlan voice quality problems, it is definitely something
different. Formerly I assumed you were trying only LTE vs LAN using
internet.

Marek


2020-06-23 15:50 GMT+02:00, Luca Bertoncello :
> Am 23.06.2020 15:43, schrieb Marek Greško:
>
> Hi
>
>>> Do you mean "my Linux-Box ignores ICMP packet unreachable" or
>>> "Deutsche
>>> Telekom ignores them"?
>>
>> I meant DT, but this was a speculation. I did not say they do. I
>> consider it highly improbable. Then I was asking whether you do. As
>> per configuration you sent you are not blocking icmp type 3 so this
>> should not be an issue.
>
> OK, so this should not be the problem...
> What can we check now?
> If you want, I can send my iptables-script. It is possible, that I have
> there an error causing this behaviour...
>
> Maybe someone in the list is an expert with iptables and can check it?
> I know this program, but I'm not really an expert...
>
> Thanks
> Luca Bertoncello
> (lucab...@lucabert.de)
>
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Re: [asterisk-users] Voice broken during calls (again...)

[Luca Bertoncello]

Am 23.06.2020 15:43, schrieb Marek Greško:

Hi

Do you mean "my Linux-Box ignores ICMP packet unreachable" or 
"Deutsche

Telekom ignores them"?


I meant DT, but this was a speculation. I did not say they do. I
consider it highly improbable. Then I was asking whether you do. As
per configuration you sent you are not blocking icmp type 3 so this
should not be an issue.


OK, so this should not be the problem...
What can we check now?
If you want, I can send my iptables-script. It is possible, that I have 
there an error causing this behaviour...


Maybe someone in the list is an expert with iptables and can check it?
I know this program, but I'm not really an expert...

Thanks
Luca Bertoncello
(lucab...@lucabert.de)

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Re: [asterisk-users] Voice broken during calls (again...)

[Marek Greško]

2020-06-23 15:02 GMT+02:00, Luca Bertoncello :
> Am 23.06.2020 14:49, schrieb Marek Greško:
>
> Hi Marek,
>
>> this could be ip address of the different interface on the same box. I
>> think it works like expected. The only exception would be if the sip
>> peer ignores the icmp packet unreachable. But I doubt this is the
>
> Do you mean "my Linux-Box ignores ICMP packet unreachable" or "Deutsche
> Telekom ignores them"?

I meant DT, but this was a speculation. I did not say they do. I
consider it highly improbable. Then I was asking whether you do. As
per configuration you sent you are not blocking icmp type 3 so this
should not be an issue.

>
>> case. Anyway you get problems also when calling to LTE phone without
>> using sip provider.
>
> I have problem calling someone outside my networks and I have problem if
> the peers are in different networks...
>
>> Let first concentrate on these calls LTE to LAN. Are you sure you do
>> not block incoming icmp unreachables? At least verify type 3 subtype 4
>> is enabled. If it is, I have no clue what is going on.
>
> Well, I limit incoming ICMP packets and I block some hosts (known
> crackers)...
> If you think, I can send you the script I use (with iptables) to manage
> my firewall, so you can check it...
> The only entries I have, having something to do with ICMP, are:
>
> --
> /bin/echo -n "Disable ICMP Redirect acceptance..."
> for f in /proc/sys/net/ipv4/conf/*/accept_redirects; do
>/bin/echo 0 > $f
> done
> /bin/echo "done."
> /sbin/iptables -A INPUT -i dsl0 -p icmp --icmp-type echo-request -m
> limit --limit 6/m --limit-burst 5 -j ACCEPT
> /sbin/iptables -A FORWARD -o dsl0 -p icmp -j ACCEPT
> --
>
> and of course other rules to allow ICMP pakets in the internal
> networks...
>
> Thanks a lot
> Luca Bertoncello
> (lucab...@lucabert.de)
>
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Re: [asterisk-users] Voice broken during calls (again...)

[Luca Bertoncello]

Am 23.06.2020 15:15, schrieb Jeff LaCoursiere:

Hi Jeff,


I have problem calling someone outside my networks and I have
problem if the peers are in different networks...


I may have missed this originally - are you saying you have trouble
when internal phones call each other, if they are on different VLAN's?
 That's a pretty big deal.


There were the results of my yesterday's tests...
If both mobile phones using SIP via LTE or both phones are in the same 
VLAN, the quality is excellent, otherwise it's bad to very bad...


But the very problem is, that all other communication between the VLANs 
don't have any problem?!?

I can transfer GB and don't have any issue...

I'm really confused...


I didn't see my post with the graphs of inter-packet latency make it
to the list (moderator?), I think the images were too large.  Recall
that clearly showed half of the packets coming inbound from DT were
*missing*, which confirms your audio experience.  I don't think that
fact has been addressed properly - it is the only smoking gun you have
so far.  If that is also happening inter-VLAN, something is seriously
wrong on the Pi.


Well, probabilly not on the PI, since, as I sayd, communication with 
both peers in the same interface work correct, but maybe my firewall 
script...



If you can reproduce this can you send me a few more packet traces,
from each of the VLAN interfaces involved?


Of course, I can do that!
Maybe I get it this evening.

Regards
Luca Bertoncello
(lucab...@lucabert.de)

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[asterisk-users] ODBC connection failure - can it be fatal?

[Antony Stone]

Hi.

I have an Asterisk 13.14.1 setup which uses ODBC to write CEL and CDR records.

The connection to my database server depends on a VPN tunnel being up, and if 
Asterisk starts before that tunnel is functional, I get messages such as the 
following in the Asterisk log file:

[2020-06-23 10:40:22.384335] delta WARNING[1697]: res_odbc.c:958 in 
odbc_obj_connect: res_odbc: Error SQLConnect=-1 errno=1045 [unixODBC]
[ma-3.0.6]Access denied for user 'Trimble'@'delta.example.net' (using 
password: YES)

The problem is that this is not a fatal error as far as Asterisk is concerned, 
so it continues to start up to the point where I get:

[2020-06-23 10:40:26.076377] delta VERBOSE[1697]: asterisk.c:4791 in 
asterisk_daemon: Asterisk Ready.

and the system then processes calls without creating any CEL or CDR entries.


Is there any way I can tell Asterisk that an ODBC connection problem is a 
fatal error, so please shut down and don't process any calls (I have an HA 
setup so another machine will handle any calls which needs processing; I just 
don't want to process calls without generating CDRs)?


Thanks,


Antony.

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Re: [asterisk-users] Voice broken during calls (again...)

[Jeff LaCoursiere]

Hi Luca,

On 6/23/20 8:02 AM, Luca Bertoncello wrote:


I have problem calling someone outside my networks and I have problem 
if the peers are in different networks...


I may have missed this originally - are you saying you have trouble when 
internal phones call each other, if they are on different VLAN's?  
That's a pretty big deal.


I didn't see my post with the graphs of inter-packet latency make it to 
the list (moderator?), I think the images were too large.  Recall that 
clearly showed half of the packets coming inbound from DT were 
*missing*, which confirms your audio experience.  I don't think that 
fact has been addressed properly - it is the only smoking gun you have 
so far.  If that is also happening inter-VLAN, something is seriously 
wrong on the Pi.


If you can reproduce this can you send me a few more packet traces, from 
each of the VLAN interfaces involved?


Always looking for real-world data to improve our tools :)

Cheers,

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Re: [asterisk-users] Voice broken during calls (again...)

[Luca Bertoncello]

Am 23.06.2020 14:49, schrieb Marek Greško:

Hi Marek,


this could be ip address of the different interface on the same box. I
think it works like expected. The only exception would be if the sip
peer ignores the icmp packet unreachable. But I doubt this is the


Do you mean "my Linux-Box ignores ICMP packet unreachable" or "Deutsche 
Telekom ignores them"?



case. Anyway you get problems also when calling to LTE phone without
using sip provider.


I have problem calling someone outside my networks and I have problem if 
the peers are in different networks...



Let first concentrate on these calls LTE to LAN. Are you sure you do
not block incoming icmp unreachables? At least verify type 3 subtype 4
is enabled. If it is, I have no clue what is going on.


Well, I limit incoming ICMP packets and I block some hosts (known 
crackers)...
If you think, I can send you the script I use (with iptables) to manage 
my firewall, so you can check it...

The only entries I have, having something to do with ICMP, are:

--
/bin/echo -n "Disable ICMP Redirect acceptance..."
for f in /proc/sys/net/ipv4/conf/*/accept_redirects; do
  /bin/echo 0 > $f
done
/bin/echo "done."
/sbin/iptables -A INPUT -i dsl0 -p icmp --icmp-type echo-request -m 
limit --limit 6/m --limit-burst 5 -j ACCEPT

/sbin/iptables -A FORWARD -o dsl0 -p icmp -j ACCEPT
--

and of course other rules to allow ICMP pakets in the internal 
networks...


Thanks a lot
Luca Bertoncello
(lucab...@lucabert.de)

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Re: [asterisk-users] Voice broken during calls (again...)

[Marek Greško]

Hello,

this could be ip address of the different interface on the same box. I
think it works like expected. The only exception would be if the sip
peer ignores the icmp packet unreachable. But I doubt this is the
case. Anyway you get problems also when calling to LTE phone without
using sip provider.

Let first concentrate on these calls LTE to LAN. Are you sure you do
not block incoming icmp unreachables? At least verify type 3 subtype 4
is enabled. If it is, I have no clue what is going on.

Marek



Marek


2020-06-23 10:11 GMT+02:00, Luca Bertoncello :
> Am 23.06.2020 10:07, schrieb Marek Greško:
>
> Hi
>
>> this is a correct response:
>>
>> From 62.156.246.57 (62.156.246.57) icmp_seq=1 Frag needed and DF set
>> (mtu = 1492)
>>
>> So PMTU discovery is working. No problem here. You got correct message
>> to lower the packet size from 62.156.246.57. This is probably the last
>> hop before your site.
>
> No, the last hop is 62.156.246.65:
>
> lucabert@ns:~$ mtr -4nr bpi.d.lucabert.com
> Start: Tue Jun 23 10:10:16 2020
> HOST: ns.lucabert.de  Loss%   Snt   Last   Avg  Best  Wrst
> StDev
>1.|-- 185.242.112.1  0.0%100.4   1.1   0.3   4.4
> 1.2
>2.|-- 84.200.230.82  0.0%100.8   0.7   0.5   0.8
> 0.0
>3.|-- 87.190.233.113 0.0%101.6   1.7   1.4   2.5
> 0.0
>4.|-- 217.5.82.940.0%107.9   7.6   7.4   7.9
> 0.0
>5.|-- 217.5.82.940.0%107.7   7.5   7.2   7.7
> 0.0
>6.|-- 62.156.246.49  0.0%107.4   7.4   7.3   7.4
> 0.0
>7.|-- 62.156.246.65  0.0%107.6   7.6   7.4   7.8
> 0.0
>8.|-- 93.241.91.232  0.0%10   21.4  21.9  21.4  24.3
> 0.7
>
> Don't know where this 62.156.246.57 comes... :(
>
> Everyway: you think, my network works as expected? At least the part
> using DSL?
> Any idea, where could be the problem?
>
> Thanks a lot
> Luca Bertoncello
> (lucab...@lucabert.de)
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
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> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
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Re: [asterisk-users] Voice broken during calls (again...)

[Administrator]

Hello

Le 23/06/2020 à 09:06, Luca Bertoncello a écrit :

Am 23.06.2020 08:43, schrieb Luca Bertoncello:

And another thing, I discovered right now...


Could you suggest me something to restrict the problem?
Currently, I think the problem can be:

1) on Asterisk
2) on my Gateway/Firewall


A couple of years ago I added this entry in my firewall:

/sbin/iptables -A FORWARD -p tcp --tcp-flags SYN,RST SYN -j TCPMSS  
--clamp-mss-to-pmtu


since I had the problem downloading data from an Internet site using 
my tablet.

I found this site explaining that:

   https://lartc.org/howto/lartc.cookbook.mtu-mss.html

I really forgot this entry, but now I checked all entries in my 
Firewall, and I see it, with my remark...

Now, the last line of the HowTo:


# iptables -A FORWARD -p tcp --tcp-flags SYN,RST SYN -j TCPMSS 
--set-mss 128


This sets the MSS of passing SYN packets to 128. Use this if you have 
VoIP with tiny packets, and huge http packets which are causing 
chopping in your voice calls.



Could it be the problem? Right now I'm not at home, so I cannot test 
it, but maybe I can add an entry like:


iptables -A FORWARD -p tcp -m multiport --ports 5060,SIP> --tcp-flags SYN,RST SYN -j TCPMSS --set-mss 128


and change the previous entry like:

iptables -A FORWARD -p tcp -i intlan0 --tcp-flags SYN,RST SYN -j 
TCPMSS  --clamp-mss-to-pmtu


to limit the behaviour on the internal LAN...

Your opinion?

Audio has nothing to do with SIP signaling 5060 port. Look at your rtp.conf

--
Daniel

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Re: [asterisk-users] Voice broken during calls (again...)

[Luca Bertoncello]

Am 23.06.2020 10:07, schrieb Marek Greško:

Hi


this is a correct response:

From 62.156.246.57 (62.156.246.57) icmp_seq=1 Frag needed and DF set
(mtu = 1492)

So PMTU discovery is working. No problem here. You got correct message
to lower the packet size from 62.156.246.57. This is probably the last
hop before your site.


No, the last hop is 62.156.246.65:

lucabert@ns:~$ mtr -4nr bpi.d.lucabert.com
Start: Tue Jun 23 10:10:16 2020
HOST: ns.lucabert.de  Loss%   Snt   Last   Avg  Best  Wrst 
StDev
  1.|-- 185.242.112.1  0.0%100.4   1.1   0.3   4.4   
1.2
  2.|-- 84.200.230.82  0.0%100.8   0.7   0.5   0.8   
0.0
  3.|-- 87.190.233.113 0.0%101.6   1.7   1.4   2.5   
0.0
  4.|-- 217.5.82.940.0%107.9   7.6   7.4   7.9   
0.0
  5.|-- 217.5.82.940.0%107.7   7.5   7.2   7.7   
0.0
  6.|-- 62.156.246.49  0.0%107.4   7.4   7.3   7.4   
0.0
  7.|-- 62.156.246.65  0.0%107.6   7.6   7.4   7.8   
0.0
  8.|-- 93.241.91.232  0.0%10   21.4  21.9  21.4  24.3   
0.7


Don't know where this 62.156.246.57 comes... :(

Everyway: you think, my network works as expected? At least the part 
using DSL?

Any idea, where could be the problem?

Thanks a lot
Luca Bertoncello
(lucab...@lucabert.de)

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Re: [asterisk-users] Voice broken during calls (again...)

[Marek Greško]

Hello,

this is a correct response:

From 62.156.246.57 (62.156.246.57) icmp_seq=1 Frag needed and DF set
(mtu = 1492)

So PMTU discovery is working. No problem here. You got correct message
to lower the packet size from 62.156.246.57. This is probably the last
hop before your site.

Marek



2020-06-23 9:40 GMT+02:00, Luca Bertoncello :
> Am 23.06.2020 09:28, schrieb Marek Greško:
>
> Hi
>
>> if you need clampmss then it is highly probable there is a PMTU
>> discovery problem. The clampmss does not work for UDP.
>
> Is there a way to check if I have this problem?
>
>> I probably counted the size incorrectly. So you are able to ping with
>> size 1464 and not with 1466. How about trying same ping sizes from the
>> internet towards your site? I mean trying to ping from sites with
>> higher MTU than yours without lower MTU links in the path.
>
> lucabert@ns:~$ ping -4 -M  do -s 1465 bpi.d.lucabert.com
> PING bpi.d.lucabert.com (93.241.91.232) 1465(1493) bytes of data.
>  From 62.156.246.57 (62.156.246.57) icmp_seq=1 Frag needed and DF set
> (mtu = 1492)
> ping: local error: Message too long, mtu=1492
> ping: local error: Message too long, mtu=1492
> ping: local error: Message too long, mtu=1492
> ^C
> --- bpi.d.lucabert.com ping statistics ---
> 4 packets transmitted, 0 received, +4 errors, 100% packet loss, time
> 3965ms
> pipe 2
>
> With paket size of 1464 it works...
>
>> You know MTU is a size of l2 frame, so using ipv6 you are able to use
>> higher payload sizes because of ip header size.
>
> OK, thanks!
> Luca Bertoncello
> (lucab...@lucabert.de)
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
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>http://lists.digium.com/mailman/listinfo/asterisk-users

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Re: [asterisk-users] Voice broken during calls (again...)

[Luca Bertoncello]

Am 23.06.2020 09:28, schrieb Marek Greško:

Hi


if you need clampmss then it is highly probable there is a PMTU
discovery problem. The clampmss does not work for UDP.


Is there a way to check if I have this problem?


I probably counted the size incorrectly. So you are able to ping with
size 1464 and not with 1466. How about trying same ping sizes from the
internet towards your site? I mean trying to ping from sites with
higher MTU than yours without lower MTU links in the path.


lucabert@ns:~$ ping -4 -M  do -s 1465 bpi.d.lucabert.com
PING bpi.d.lucabert.com (93.241.91.232) 1465(1493) bytes of data.
From 62.156.246.57 (62.156.246.57) icmp_seq=1 Frag needed and DF set 
(mtu = 1492)

ping: local error: Message too long, mtu=1492
ping: local error: Message too long, mtu=1492
ping: local error: Message too long, mtu=1492
^C
--- bpi.d.lucabert.com ping statistics ---
4 packets transmitted, 0 received, +4 errors, 100% packet loss, time 
3965ms

pipe 2

With paket size of 1464 it works...


You know MTU is a size of l2 frame, so using ipv6 you are able to use
higher payload sizes because of ip header size.


OK, thanks!
Luca Bertoncello
(lucab...@lucabert.de)

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Re: [asterisk-users] Voice broken during calls (again...)

[Marek Greško]

Hello,

if you need clampmss then it is highly probable there is a PMTU
discovery problem. The clampmss does not work for UDP.

I probably counted the size incorrectly. So you are able to ping with
size 1464 and not with 1466. How about trying same ping sizes from the
internet towards your site? I mean trying to ping from sites with
higher MTU than yours without lower MTU links in the path.

You know MTU is a size of l2 frame, so using ipv6 you are able to use
higher payload sizes because of ip header size.

Marek


2020-06-23 9:06 GMT+02:00, Luca Bertoncello :
> Am 23.06.2020 08:43, schrieb Luca Bertoncello:
>
> And another thing, I discovered right now...
>
>> Could you suggest me something to restrict the problem?
>> Currently, I think the problem can be:
>>
>> 1) on Asterisk
>> 2) on my Gateway/Firewall
>
> A couple of years ago I added this entry in my firewall:
>
> /sbin/iptables -A FORWARD -p tcp --tcp-flags SYN,RST SYN -j TCPMSS
> --clamp-mss-to-pmtu
>
> since I had the problem downloading data from an Internet site using my
> tablet.
> I found this site explaining that:
>
> https://lartc.org/howto/lartc.cookbook.mtu-mss.html
>
> I really forgot this entry, but now I checked all entries in my
> Firewall, and I see it, with my remark...
> Now, the last line of the HowTo:
>
> 
> # iptables -A FORWARD -p tcp --tcp-flags SYN,RST SYN -j TCPMSS --set-mss
> 128
>
> This sets the MSS of passing SYN packets to 128. Use this if you have
> VoIP with tiny packets, and huge http packets which are causing chopping
> in your voice calls.
> 
>
> Could it be the problem? Right now I'm not at home, so I cannot test it,
> but maybe I can add an entry like:
>
> iptables -A FORWARD -p tcp -m multiport --ports 5060, SIP> --tcp-flags SYN,RST SYN -j TCPMSS --set-mss 128
>
> and change the previous entry like:
>
> iptables -A FORWARD -p tcp -i intlan0 --tcp-flags SYN,RST SYN -j TCPMSS
> --clamp-mss-to-pmtu
>
> to limit the behaviour on the internal LAN...
>
> Your opinion?
>
> Thanks a lot!
> Luca Bertoncello
> (lucab...@lucabert.de)
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

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Re: [asterisk-users] Voice broken during calls (again...)

[Luca Bertoncello]

Am 23.06.2020 09:19, schrieb Administrator:

Hi Daniel

Audio has nothing to do with SIP signaling 5060 port. Look at your 
rtp.conf


You're right...
I have to restrict to the ports I configured in rtp.conf...
So like:

iptables -A FORWARD -p tcp -m multiport --ports -ports 1:15100 
--tcp-flags SYN,RST SYN -j TCPMSS --set-mss 128


?

Or I just have to use:

iptables -A FORWARD -p tcp --tcp-flags SYN,RST SYN -j TCPMSS --set-mss 
128


instead of:

iptables -A FORWARD -p tcp --tcp-flags SYN,RST SYN -j TCPMSS  
--clamp-mss-to-pmtu


?

Thanks
Luca Bertoncello
(lucab...@lucabert.de)

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Re: [asterisk-users] Voice broken during calls (again...)

[Luca Bertoncello]

Am 23.06.2020 08:43, schrieb Luca Bertoncello:

And another thing, I discovered right now...


Could you suggest me something to restrict the problem?
Currently, I think the problem can be:

1) on Asterisk
2) on my Gateway/Firewall


A couple of years ago I added this entry in my firewall:

/sbin/iptables -A FORWARD -p tcp --tcp-flags SYN,RST SYN -j TCPMSS  
--clamp-mss-to-pmtu


since I had the problem downloading data from an Internet site using my 
tablet.

I found this site explaining that:

   https://lartc.org/howto/lartc.cookbook.mtu-mss.html

I really forgot this entry, but now I checked all entries in my 
Firewall, and I see it, with my remark...

Now, the last line of the HowTo:


# iptables -A FORWARD -p tcp --tcp-flags SYN,RST SYN -j TCPMSS --set-mss 
128


This sets the MSS of passing SYN packets to 128. Use this if you have 
VoIP with tiny packets, and huge http packets which are causing chopping 
in your voice calls.



Could it be the problem? Right now I'm not at home, so I cannot test it, 
but maybe I can add an entry like:


iptables -A FORWARD -p tcp -m multiport --ports 5060,SIP> --tcp-flags SYN,RST SYN -j TCPMSS --set-mss 128


and change the previous entry like:

iptables -A FORWARD -p tcp -i intlan0 --tcp-flags SYN,RST SYN -j TCPMSS  
--clamp-mss-to-pmtu


to limit the behaviour on the internal LAN...

Your opinion?

Thanks a lot!
Luca Bertoncello
(lucab...@lucabert.de)

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Re: [asterisk-users] Voice broken during calls (again...)

[Luca Bertoncello]

Am 22.06.2020 20:09, schrieb Luca Bertoncello:

A couple of other ideas...

Conclusion (maybe!): it can *not* be a problem in the DSL connection 
and

*maybe* it is not a problem in the communication with the Server of
Deutsche Telekom, since I have many problems to communicate between two
peers in local Asterisk if one is over LTE and the other in local LAN
(but curiously *not* if both peers are in local LAN or both via LTE).


I think, the problem with bad quality and broken voice just happens if 
the peers are in different LANs, since if I call my wife's phone (VLAN 
"phone") using my mobile phone via SIP (in VLAN "intlan") the quality is 
bad, but if I call her using my phone in VLAN "phone" or if both peers 
use SIP via LTE the quality is very good...


Could you suggest me something to restrict the problem?
Currently, I think the problem can be:

1) on Asterisk
2) on my Gateway/Firewall

At home I have many VLANs, that normally *not* communicate together 
(some exceptions are of course implemented). The phones don't reach the 
Internet via NAT (VLAN "phone" has no routing in Internet).

The mobile phones are in VLAN "intlan", with routing in Internet.

Any idea?

Thanks
Luca Bertoncello
(lucab...@lucabert.de)

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Re: [asterisk-users] Voice broken during calls (again...)

[Luca Bertoncello]

Am 23.06.2020 07:27, schrieb Luca Bertoncello:

I again


Do not change MTU. Probably there will be another problem. I expect
packet size 1466 would pass and higher will have the same result. It


I checked it, and I see, that the maximum I can use is a paket size of 
1464 with all hosts via IPv4.

Via IPv6 I can use higher MTU, but I really can't explain why...

Can someone explain me what does it mean, if this is a problem for VoIP 
and how I can solve it?


Thanks
Luca Bertoncello
(lucab...@lucabert.de)

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