Re: [OpenSIPS-Users] Opensips switch from SQL to NoSQL

[Eric Tamme]

Not all cachedb backends have implementations that allow you to use all 
modules.   The db api exports specific functions that may not be 
implemented for a given cache db module.  You have found an example of 
one where a module requires a db api function implementation, and the 
cachedb module you are using does not implement it.

-Eric

On 07/14/2016 11:31 AM, feroze waris wrote:

Hi Daniel and Benajmin,

Thank you for reply and sharing link of mongodb basics. Yes there is a 
performance issue. I want that opensips writes and get information of 
dialog, location, subscriber etc from cachedb(mongodb). For this task 
i came across with the module named *db_cachedb *( 
http://www.opensips.org/html/docs/modules/2.1.x/db_cachedb.html ). *


*
 According to documentation what this module do is " *The db_cachedb 
module will expose the same front db api, however it will run on top 
of a NoSQL back-end, emulating the SQL calls to the back-end specific 
queries*."


I implemented this module and also created collections in mongodb 
according to mysql tables list. Now when i start opensips it gives me 
following error


Jul 14 17:27:45 localhost [23610]: ERROR:core:db_check_api: module 
db_mongodb does not export db_use_table function
Jul 14 17:27:45 localhost [23610]: ERROR:uri:mod_init: No database 
module found
Jul 14 17:27:45 localhost [23610]: ERROR:core:init_mod: failed to 
initialize module uri
Jul 14 17:27:45 localhost [23610]: ERROR:core:main: error while 
initializing modules


my opensips version is 2.1.3


On Wed, Jul 13, 2016 at 6:52 PM, Benjamin Cropley 
> wrote:


You don't have to explicitly define a schema when you insert a
document into MongoDB.

For example, you could create a MongoDB database, and then
immediately do an insert into an insert into a collection without
even creating it or a 'schema'.

Maybe you should do some reading about MongoDB and NoSQL in
general, as a basic understanding should have answered that
question :)

https://docs.mongodb.com/manual/faq/fundamentals/

On Wed, Jul 13, 2016 at 2:44 PM, feroze waris
> wrote:

Hi,

Can anyone tell me how to move opensips DB from SQL to NoSQL.
I have seen a module name db_cachedb which has a support for
mongodb but how to move mysql schema to mongodb schema.

Regards
Feroze

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07539 366 905

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Re: [OpenSIPS-Users] horizontal scaling / dimensioning

[Eric Tamme]

Short answer,  no.  There are many variables that will change what a 
system is capable of and without understanding all the pieces involved, 
you can not make any estimate at sizing.


On 07/13/2016 11:51 AM, Owais Ahmad wrote:

Hi all,

Is there a way I can theoretically determine the max number of dialogs 
and cps a system of known specifications can handle?


Also share useful tools for dimensioning, and your findings of max 
concurrent calls and cps on single opensips instances with specs.


Thanks,
Owais


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Re: [OpenSIPS-Users] Number validation

[Eric Tamme]

yes, https://www.itu.int/rec/T-REC-E.164-201011-I/en


On 07/08/2016 07:05 AM, Hristo Donev wrote:

Hello,

I thinking of performing number validation before I pass the request 
to the upstream provider in order to minimize capacity waste because 
of malformed numbers. Can this be something I can do in OpenSIPS?
Also, any idea of  good source for this validation for global 
(international) numbers?


Best regards,
Hristo



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Re: [OpenSIPS-Users] Weird behaviour when replying to an OPTIONS with a Cseq of 0

[Eric Tamme]

obscuring the ip/hostnames in this makes it difficult to troubleshoot - 
please dont do that in the future.


I am not certain the CSeq is the issue.  I believe that the contact is 
broken, aka it contains localhost, so when opensips tries to route it, 
it fails on udp_send: sendto.


Again, this is my guess, but since you have manipulated and obscured the 
complete sip trace it makes it difficult to confirm.


-Eric

On 07/06/2016 10:44 AM, Karolis Pabijanskas wrote:

Hi List,

We seem to be hitting a strange behaviour when we get an OPTIONS ping 
with a Cseq of 0. (latest 1.11 branch).


Our routing script contains this at the very beginning to decline 
OPTIONS messages:

route {
if (is_method("OPTIONS")) {
sl_send_reply("501", "Method not allowed");
exit;
};
   ## blah...
}

If we send this OPTIONS request:

2016-07-06 17:12:05 +0100 : CLIENT_IP:5061 -> OPENSIPS_IP:5060
OPTIONSsip:200@HOSTNAME:5060 SIP/2.0 Via: SIP/2.0/UDP 
127.0.0.1:5061;branch=z9hG4bK-895-1-0 From: sipp 
;tag=1To:  Call-ID: 
1-895@127.0.0.1CSeq: 0 OPTIONSContact: 
sip:100@127.0.0.1:5061Max-Forwards: 100 Content-Length: 0


There is no reply from OpenSIPS. Interestingly, siptrace module is 
also running and saving captures in Homer. Homer, actually, is getting 
a copy of the generated reply:

2016-07-06 17:12:05 +0100 : OPENSIPS_IP:5060 -> CLIENT_IP:5061
SIP/2.0 501 Method not allowed Via: SIP/2.0/UDP 
127.0.0.1:5061;received=*CLIENT_IP*;branch=z9hG4bK-895-1-0 From: sipp 
;tag=1To: 
;tag=06a366df8881a48001f15f72f7138d9f.7522 
Call-ID: 1-895@127.0.0.1CSeq: 0 OPTIONS Server: User Agent String 
Content-Length: 0
But running a tcpdump on the OpenSIPS host reveals that no actual 
packet is ever sent to the client. Debug shows:
Jul  6 17:40:25 HOSTNAME /sbin/opensips[48357]: ERROR:core:udp_send: 
sendto(sock,0x7f7867aee470,324,0,0x7fff77d00090,16): Operation not 
permitted(1)
Jul  6 17:40:25 HOSTNAME /sbin/opensips[48357]: ERROR:sl:msg_send: 
udp_send failed


OpenSIPS is running as root.

Switching Cseq to 1 in that original OPTIONS message works. But in 
this particular case we have no control over the Cseq of the host we 
are being pinged from, and need to reply. According to the RFC, Cseq 
should be a 32bit unsigned integer, so 0 should not be an issue.


Any ideas?

Thanks!
Karolis



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Re: [OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc

[Eric Tamme]

And are you forcing RTPengine to act as an ice light client?  It looks 
like you are gettin a single ICE candidate in the answer back from 
freeswitch which would indicate that you are.


I'd check your chrome webrtc statistics to see if tis failed to do do 
ice/stun negotiation on the 183.  In general the signalling looks good.


I think you may have an error on your Freeswitch side - some thing that 
is trying to force it to use SRTP all the time, even though the 
signalling has requested plain RTP (to freeswitch).


I think you should ask in #freeswitch on freenode at this point.

-Eric

On 06/23/2016 01:42 PM, John Nash wrote:
Actually the issue is i hear no audio on either side and just after 
session progress (I guess when media starts coming from remote media 
server) i see error  "SRTP output wanted, but no crypto suite was 
negotiated"


I had also checked media logs i could see RTP packets being sent from 
freeswitch to RTPengine IP but there was no packet at all just after 
that. Ideally after RTP packet from freeswitch to rtpengine, Rtpengine 
should send that packet to browser using wss?


On Fri, Jun 24, 2016 at 1:05 AM, Eric Tamme <e...@uphreak.com 
<mailto:e...@uphreak.com>> wrote:


So - i dont see a problem here - Chrome is getting
UDP/TLS/RTP/SAVPF and Freeswitch is getting RTP/AVP.  Freeswitch
responded to the offer in the invite with an answer in the 183,
and in the 200.  What is the failure you are seeing, and where is
it happening (in freeswitch? in the browser?)

The only thing that looks bad is that you are retransmitting the
ACK which FS either ... doesnt like, or is never getting,  because
it keeps retransmitting the 200, which is why you get a 481 when
you send BYE.

-Eric


On 06/23/2016 01:24 PM, John Nash wrote:

OK here is the log
https://gist.github.com/johnnash13/0d2cb5238f3551cd3a8c6b4e638dd744

Sorry took me a while to convert wireshark trace to text file.

My freeswitch is running on private IP (127.0.0.1) and opensips I
run on both public and private so that for outside world opensips
is the only public IP they see. In proxy log I pasted Opensips
===> Freeswitch logs and back.






On Fri, Jun 24, 2016 at 12:43 AM, Eric Tamme <e...@uphreak.com
<mailto:e...@uphreak.com>> wrote:

No - it's annoying to look at a trace that's had information
removed and try and piece together whats happening.  Your
paranoid side is wrong, sorry.

-Eric


On 06/23/2016 01:06 PM, Patrick Wakano wrote:

my paranoic side would recommend to hide/change private
informations, specially any authentication line that might
appear... this is certainly a sort of social engineering
threat we should worry...
better be safe than sorry


On Thu, Jun 23, 2016 at 3:31 PM, Eric Tamme
<e...@uphreak.com <mailto:e...@uphreak.com>> wrote:

I mean you can use a private gist, but you will be
publishing the link in a public email list. In general I
personally dont believe revealing ip addresses etc. is
any problem - to put my money where my mouth is here is
a gist link to an unaltered SIP trace on my server :)

https://gist.github.com/etamme/b864010448a29007b7e0457682e81d52

-Eric


On 06/23/2016 12:23 PM, John Nash wrote:

Ok i am ready with logs. About gist may I use private
option as traces have our IPs, user

    On Thu, Jun 23, 2016 at 10:32 PM, Eric Tamme
<e...@uphreak.com <mailto:e...@uphreak.com>> wrote:

Hey John,

Please paste a full UNALTERED sip trace into a gist
(gist.github.com <http://gist.github.com>) from the
proxy servers perspective and provide a link so
that we can see what comes in, and what goes out
from both sides.

EG: ngrep -qtd any -W byline port 5060

This will show us the traffic that is leaving the
proxy destined for the Freeswitch box, and what the
freeswitch box sends back.

Also - you can look in your browsers console log
and provide the SIP trace from there in a seperate
gist, so that we can see what opensips sends back
up to your browser.

-Eric



Am I using correct sip.js example? I copied it to
my server and accessing it using https: (used
letsencrypt)

On Thu, Jun 23, 2016 at 7:58 PM, Eric Tamme
<e...@uphreak.com <mailto:e...@uphreak.com>> wrote:

1. I would suggest using SIP.js -
https://github.com/onsip/SIP.js it is a much

Re: [OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc

[Eric Tamme]

So - i dont see a problem here - Chrome is getting UDP/TLS/RTP/SAVPF and 
Freeswitch is getting RTP/AVP.  Freeswitch responded to the offer in the 
invite with an answer in the 183, and in the 200.  What is the failure 
you are seeing, and where is it happening (in freeswitch? in the browser?)


The only thing that looks bad is that you are retransmitting the ACK 
which FS either ... doesnt like, or is never getting,  because it keeps 
retransmitting the 200, which is why you get a 481 when you send BYE.


-Eric

On 06/23/2016 01:24 PM, John Nash wrote:
OK here is the log 
https://gist.github.com/johnnash13/0d2cb5238f3551cd3a8c6b4e638dd744


Sorry took me a while to convert wireshark trace to text file.

My freeswitch is running on private IP (127.0.0.1) and opensips I run 
on both public and private so that for outside world opensips is the 
only public IP they see. In proxy log I pasted Opensips ===> 
Freeswitch logs and back.







On Fri, Jun 24, 2016 at 12:43 AM, Eric Tamme <e...@uphreak.com 
<mailto:e...@uphreak.com>> wrote:


No - it's annoying to look at a trace that's had information
removed and try and piece together whats happening.  Your paranoid
side is wrong, sorry.

-Eric


On 06/23/2016 01:06 PM, Patrick Wakano wrote:

my paranoic side would recommend to hide/change private
informations, specially any authentication line that might
appear... this is certainly a sort of social engineering threat
we should worry...
better be safe than sorry


On Thu, Jun 23, 2016 at 3:31 PM, Eric Tamme <e...@uphreak.com
<mailto:e...@uphreak.com>> wrote:

I mean you can use a private gist, but you will be publishing
the link in a public email list. In general I personally dont
believe revealing ip addresses etc. is any problem - to put
my money where my mouth is here is a gist link to an
unaltered SIP trace on my server :)

https://gist.github.com/etamme/b864010448a29007b7e0457682e81d52

-Eric


On 06/23/2016 12:23 PM, John Nash wrote:

Ok i am ready with logs. About gist may I use private option
as traces have our IPs, user

On Thu, Jun 23, 2016 at 10:32 PM, Eric Tamme
<e...@uphreak.com <mailto:e...@uphreak.com>> wrote:

Hey John,

Please paste a full UNALTERED sip trace into a gist
(gist.github.com <http://gist.github.com>) from the
proxy servers perspective and provide a link so that we
can see what comes in, and what goes out from both sides.

EG: ngrep -qtd any -W byline port 5060

This will show us the traffic that is leaving the proxy
destined for the Freeswitch box, and what the freeswitch
box sends back.

Also - you can look in your browsers console log and
provide the SIP trace from there in a seperate gist, so
that we can see what opensips sends back up to your browser.

-Eric



Am I using correct sip.js example? I copied it to my
server and accessing it using https: (used letsencrypt)

    On Thu, Jun 23, 2016 at 7:58 PM, Eric Tamme
<e...@uphreak.com <mailto:e...@uphreak.com>> wrote:

1. I would suggest using SIP.js -
https://github.com/onsip/SIP.js it is a much more
active project that sipml5.

2. Im guessing that you are not properly passing
flags to RTPEngine.  If you want to have DTLS-SRTP
between the browser, and plain RTP/AVP between
RTPEngine and freeswitch, you need to "offer"
rtp/avp to freeswitch, and "answer" dtls-srtp back
up to the browser.

the offer to freeswitch would be:

 $var(rtpengine_flags) = "RTP/AVP replace-session-connection 
replace-origin ICE=remove";

and the answer back up to the browswer would be:

 $var(rtpengine_flags) = "UDP/TLS/RTP/SAVPF ICE=force";


-Eric



On 06/23/2016 08:20 AM, John Nash wrote:

I am following
http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2
and trying to test a call

sipml5 --->Opensips + rtpengine >
SIP end point (Freeswitch)

But I do not have any audio on both sides. I see
this error at rtpengine log "SRTP output wanted,
but no crypto suite was negotiated"

Anyone tested this scenario positive?


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Re: [OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc

[Eric Tamme]

No - it's annoying to look at a trace that's had information removed and 
try and piece together whats happening.  Your paranoid side is wrong, sorry.


-Eric

On 06/23/2016 01:06 PM, Patrick Wakano wrote:
my paranoic side would recommend to hide/change private informations, 
specially any authentication line that might appear... this is 
certainly a sort of social engineering threat we should worry...

better be safe than sorry


On Thu, Jun 23, 2016 at 3:31 PM, Eric Tamme <e...@uphreak.com 
<mailto:e...@uphreak.com>> wrote:


I mean you can use a private gist, but you will be publishing the
link in a public email list.  In general I personally dont believe
revealing ip addresses etc. is any problem - to put my money where
my mouth is here is a gist link to an unaltered SIP trace on my
server :)

https://gist.github.com/etamme/b864010448a29007b7e0457682e81d52

-Eric


On 06/23/2016 12:23 PM, John Nash wrote:

Ok i am ready with logs. About gist may I use private option as
traces have our IPs, user

On Thu, Jun 23, 2016 at 10:32 PM, Eric Tamme <e...@uphreak.com
<mailto:e...@uphreak.com>> wrote:

Hey John,

Please paste a full UNALTERED sip trace into a gist
(gist.github.com <http://gist.github.com>) from the proxy
servers perspective and provide a link so that we can see
what comes in, and what goes out from both sides.

EG: ngrep -qtd any -W byline port 5060

This will show us the traffic that is leaving the proxy
destined for the Freeswitch box, and what the freeswitch box
sends back.

Also - you can look in your browsers console log and provide
the SIP trace from there in a seperate gist, so that we can
see what opensips sends back up to your browser.

-Eric



Am I using correct sip.js example? I copied it to my server
and accessing it using https: (used letsencrypt)

On Thu, Jun 23, 2016 at 7:58 PM, Eric Tamme
<e...@uphreak.com <mailto:e...@uphreak.com>> wrote:

1. I would suggest using SIP.js -
https://github.com/onsip/SIP.js it is a much more active
project that sipml5.

2. Im guessing that you are not properly passing flags
to RTPEngine.  If you want to have DTLS-SRTP between the
browser, and plain RTP/AVP between RTPEngine and
freeswitch, you need to "offer" rtp/avp to freeswitch,
and "answer" dtls-srtp back up to the browser.

the offer to freeswitch would be:

 $var(rtpengine_flags) = "RTP/AVP replace-session-connection 
replace-origin ICE=remove";

and the answer back up to the browswer would be:

 $var(rtpengine_flags) = "UDP/TLS/RTP/SAVPF ICE=force";


-Eric



On 06/23/2016 08:20 AM, John Nash wrote:

I am following
http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2
and trying to test a call

sipml5 --->Opensips + rtpengine > SIP
end point (Freeswitch)

But I do not have any audio on both sides. I see this
error at rtpengine log "SRTP output wanted, but no
crypto suite was negotiated"

Anyone tested this scenario positive?


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Re: [OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc

[Eric Tamme]

I mean you can use a private gist, but you will be publishing the link 
in a public email list.  In general I personally dont believe revealing 
ip addresses etc. is any problem - to put my money where my mouth is 
here is a gist link to an unaltered SIP trace on my server :)


https://gist.github.com/etamme/b864010448a29007b7e0457682e81d52

-Eric

On 06/23/2016 12:23 PM, John Nash wrote:
Ok i am ready with logs. About gist may I use private option as traces 
have our IPs, user


On Thu, Jun 23, 2016 at 10:32 PM, Eric Tamme <e...@uphreak.com 
<mailto:e...@uphreak.com>> wrote:


Hey John,

Please paste a full UNALTERED sip trace into a gist
(gist.github.com <http://gist.github.com>) from the proxy servers
perspective and provide a link so that we can see what comes in,
and what goes out from both sides.

EG: ngrep -qtd any -W byline port 5060

This will show us the traffic that is leaving the proxy destined
for the Freeswitch box, and what the freeswitch box sends back.

Also - you can look in your browsers console log and provide the
SIP trace from there in a seperate gist, so that we can see what
opensips sends back up to your browser.

-Eric



Am I using correct sip.js example? I copied it to my server and
accessing it using https: (used letsencrypt)

On Thu, Jun 23, 2016 at 7:58 PM, Eric Tamme <e...@uphreak.com
<mailto:e...@uphreak.com>> wrote:

1. I would suggest using SIP.js -
https://github.com/onsip/SIP.js it is a much more active
project that sipml5.

2. Im guessing that you are not properly passing flags to
RTPEngine.  If you want to have DTLS-SRTP between the
browser, and plain RTP/AVP between RTPEngine and freeswitch,
you need to "offer" rtp/avp to freeswitch, and "answer"
dtls-srtp back up to the browser.

the offer to freeswitch would be:

 $var(rtpengine_flags) = "RTP/AVP replace-session-connection 
replace-origin ICE=remove";

and the answer back up to the browswer would be:

 $var(rtpengine_flags) = "UDP/TLS/RTP/SAVPF ICE=force";


-Eric



On 06/23/2016 08:20 AM, John Nash wrote:

I am following
http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2 and
trying to test a call

sipml5 --->Opensips + rtpengine > SIP end
point (Freeswitch)

But I do not have any audio on both sides. I see this error
at rtpengine log "SRTP output wanted, but no crypto suite
was negotiated"

Anyone tested this scenario positive?


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Re: [OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc

[Eric Tamme]

Hey John,

Please paste a full UNALTERED sip trace into a gist (gist.github.com) 
from the proxy servers perspective and provide a link so that we can see 
what comes in, and what goes out from both sides.


EG: ngrep -qtd any -W byline port 5060

This will show us the traffic that is leaving the proxy destined for the 
Freeswitch box, and what the freeswitch box sends back.


Also - you can look in your browsers console log and provide the SIP 
trace from there in a seperate gist, so that we can see what opensips 
sends back up to your browser.


-Eric

Am I using correct sip.js example? I copied it to my server and 
accessing it using https: (used letsencrypt)


On Thu, Jun 23, 2016 at 7:58 PM, Eric Tamme <e...@uphreak.com 
<mailto:e...@uphreak.com>> wrote:


1. I would suggest using SIP.js - https://github.com/onsip/SIP.js
it is a much more active project that sipml5.

2. Im guessing that you are not properly passing flags to
RTPEngine.  If you want to have DTLS-SRTP between the browser, and
plain RTP/AVP between RTPEngine and freeswitch, you need to
"offer" rtp/avp to freeswitch, and "answer" dtls-srtp back up to
the browser.

the offer to freeswitch would be:

 $var(rtpengine_flags) = "RTP/AVP replace-session-connection 
replace-origin ICE=remove";

and the answer back up to the browswer would be:

 $var(rtpengine_flags) = "UDP/TLS/RTP/SAVPF ICE=force";


-Eric



On 06/23/2016 08:20 AM, John Nash wrote:

I am following
http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2 and
trying to test a call

sipml5 --->Opensips + rtpengine > SIP end point
(Freeswitch)

But I do not have any audio on both sides. I see this error at
rtpengine log "SRTP output wanted, but no crypto suite was
negotiated"

Anyone tested this scenario positive?


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Re: [OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc

[Eric Tamme]

1. I would suggest using SIP.js - https://github.com/onsip/SIP.js it is 
a much more active project that sipml5.


2. Im guessing that you are not properly passing flags to RTPEngine.  If 
you want to have DTLS-SRTP between the browser, and plain RTP/AVP 
between RTPEngine and freeswitch, you need to "offer" rtp/avp to 
freeswitch, and "answer" dtls-srtp back up to the browser.


the offer to freeswitch would be:

$var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin 
ICE=remove";

and the answer back up to the browswer would be:

$var(rtpengine_flags) = "UDP/TLS/RTP/SAVPF ICE=force";


-Eric


On 06/23/2016 08:20 AM, John Nash wrote:
I am following 
http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2 and 
trying to test a call


sipml5 --->Opensips + rtpengine > SIP end point 
(Freeswitch)


But I do not have any audio on both sides. I see this error at 
rtpengine log "SRTP output wanted, but no crypto suite was negotiated"


Anyone tested this scenario positive?


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Re: [OpenSIPS-Users] What is the best SQLite version to be used with OpenSIPS 2.2? We investigate a memory leak.

[Eric Tamme]

Hey Rodrigo,

Are you running https://github.com/etamme/federated-sip by chance? Your 
use of the PCRE module made me think you might be.  I run federated-sip 
and I do use sqlite3 with opensips - my current sqlite version is:  
sqlite-3.7.17-4.el7.x86_64


I do not know that I have memory leaks outside of what I reported in the 
github issue.


-Eric

On 06/17/2016 11:08 AM, Rodrigo Pimenta Carvalho wrote:


Thank you Ionut.


We will try it so.


Today morning, we noticed that OpenSIPS 2.2 while running and using 
SQLite, without online clients, without registers and without calls, 
causes a memory leak. That is, OpenSIPS even without any SIP request 
causes a memory leak due to the use of SQLite.



After updating the SQLite to a new version, such memory leak was vanished.


However, even with the newest SQLite, we still get memory leaks again 
if the proxy receives SIP REGISTER messages. That is, we get the issue 
every time some client registers. In this case we saw the memory leak 
in : " modparam("db_sqlite", "load_extension", 
"/usr/lib/sqlite3/pcre.so")"



Let us try the new solution and see what happens.


Best regards!


RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979



*De:* users-boun...@lists.opensips.org 
 em nome de Ionut Ionita 


*Enviado:* sexta-feira, 17 de junho de 2016 11:45
*Para:* OpenSIPS users mailling list
*Assunto:* Re: [OpenSIPS-Users] What is the best SQLite version to be 
used with OpenSIPS 2.2? We investigate a memory leak.

Hi Rodrigo,

Pushed a fix both into 2.2[0] and master[1] branches. If you still 
think sqlite leaks even with this fix,

please feel free to open an issue on github.

[0] 
https://github.com/OpenSIPS/opensips/commit/c1aa55e73f7c56ea4894997aeb7f25a03cb289cf 




[sqlite][bugfix] free column names when freeing the result · 
OpenSIPS/opensips@c1aa55e 


github.com
(cherry picked from commit cf380773cec9f91ad08e343c03261154dabc36a5)


[1] 
https://github.com/OpenSIPS/opensips/commit/c1aa55e73f7c56ea4894997aeb7f25a03cb289cf 


Regrads,
Ionut Ionita
OpenSIPS Developer
On 06/17/2016 05:19 PM, Rodrigo Pimenta Carvalho wrote:


Hi Liviu.


Very good.


We will see the resolution process.

Thank you very much!

Regards.


RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979



*De:* users-boun...@lists.opensips.org 
 em nome de Liviu Chircu 


*Enviado:* sexta-feira, 17 de junho de 2016 11:14
*Para:* users@lists.opensips.org
*Assunto:* Re: [OpenSIPS-Users] What is the best SQLite version to be 
used with OpenSIPS 2.2? We investigate a memory leak.

Hi Rodrigo!

A GitHub issue [1] regarding this leak was just reported today by 
Eric, so you can track the resolution process over there! You can 
even subscribe to that ticket if you have an account, in order to 
receive emails.


[1]: https://github.com/OpenSIPS/opensips/issues/911


2.2 runs out of pkg_mem because of db/db_res.c memory leak · Issue 
#911 · OpenSIPS/opensips 


github.com
OpenSIPS 2.2 will run out of pkg_mem, i believe because of a leak in 
db/db_res.c Full memlog dump is available here: 
https://gist.github.com/etamme/7d42024ad684fe834b9fd514d2bd2412 I am 
using...



Liviu Chircu
OpenSIPS Developer
http://www.opensips-solutions.com
On 17.06.2016 17:02, Rodrigo Pimenta Carvalho wrote:


Hi.


People from my team is investigating a memory leak related to 
OpenSIPS 2.2.



As I had commented in another discussion in the past, it seems that 
the problem comes from SQLite we are using as the Registrar for our 
OpenSIPS 2.2.


For example, a script opensips.cfg that doesn't use SQLite didn't 
cause memory leak. But, a script that uses it and use another module 
that needs a database (EX: auth.so) causes memory leak.



We are still in the beginning of the investigation.

So, what is the best version of SQLite to be used with OpenSIPS 2.2? 
That is, what version of SQLite was very well tested with OpenSIPS 
2.2 and worked  without memory leaks or others issues?



Any suggestion will be very helpful!


Best regards.


RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979


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Re: [OpenSIPS-Users] .Net Integartion To OpenSips

[Eric Tamme]

I mean, opensipsctl add just inserts to a database, so you could avoid 
using the opensipsctl command line client all together and just insert 
directly to the database.



On 03/01/2016 09:10 AM, Adrian Newell wrote:


My terminology was not good,  what I meant was is there any way that I 
can call the ‘opensisctl add’ command line function to create a user 
account remotely on the SIPS server, such as from a .Net application.


I need to interface into OpenSIPS to create the user accounts.   If 
the OpenSIPS Control Panel has this functionality I would be 
interested in knowing how it does it.


Thanks



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Re: [OpenSIPS-Users] wss support in opensips.

[Eric Tamme]

yes it is on master



On Mar 5, 2016, 3:25 AM, at 3:25 AM, johan  wrote:
>Dears,
>
>is wss support already there in opensips devel branch ?
>
>--
>
>
>
>
>
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Re: [OpenSIPS-Users] .Net Integartion To OpenSips

[Eric Tamme]

Use a .net SIP library.

On 02/26/2016 05:57 AM, Adrian Newell wrote:


Any suggestions about the best way to register a phone with OpenSIPS 
from a .Net client application ?   You don’t appear to have an API 
that can be used from .Net,  would it be easier to write directly into 
the database ?


I can only find one third party on the internet who do provide a .Net 
API but would prefer to understand your stance on this.


*Kind Regards*

*Adrian Newell *

.



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Re: [OpenSIPS-Users] What does exactally mean lots of warnings about timer_ticker in the log (OpenSIPS 2.1)?

[Eric Tamme]

Rodrigo,

I believe if you create partitions for your nathelper call backs see:

http://www.opensips.org/html/docs/modules/2.1.x/nathelper.html#id293468

You will not have this issue any more.  The nathelper callback is taking 
a long time to ping all the nat'd contacts most likely, so it doesnt 
finish its job before its time to run again.  If you break up the 
pinging into multiple partitions it will go faster, and prevent the 
overlap from happening.


-Eric

On 02/23/2016 02:22 PM, Eric Tamme wrote:

Hey again Rodrigo,

I had further discussions with Bogdan which I will summarize here in 
an attempt to clarify.


Because the new 2.x system is implemented as a reactor, the 
implementation of timer based callbacks has changed.  There is a 
single timer process who's only job is to keep track of jobs that are 
to be scheduled, when it decides a job/callback needs to happen 
because of the current time, it writes to a shared pipe which is read 
by ALL other processes of the reactor.  Any reactor process that is 
free, will take this callback off the pipe and run it.  In this way, 
there is no way to actually block the timer process.


However, what can happen, and what I believe you are seeing here is 
that the timer process has not finished going through the list of 
timer handlers and it should have started processing the list again 
already - so the timer list scan is overlapping.


I'm not 100% sure how this can happen, perhaps you have a very long 
timer list and it is causing the timer process to fail to complete 
before its next scheduled iteration - either than or it is not able to 
write to the shared pipe for some reason.


I would bring the question to either IRC, or to the devel list with as 
much detail as possible.


-Eric


On 02/23/2016 01:59 PM, Rodrigo Pimenta Carvalho wrote:


Ok Eric.


Thank you very much.

I will analyze my OpenSIPS configuration and try to discover what 
action is causing the block situation.



Best regards.


RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979



*De:* users-boun...@lists.opensips.org 
<users-boun...@lists.opensips.org> em nome de Eric Tamme 
<e...@uphreak.com>

*Enviado:* terça-feira, 23 de fevereiro de 2016 17:51
*Para:* OpenSIPS users mailling list
*Assunto:* Re: [OpenSIPS-Users] What does exactally mean lots of 
warnings about timer_ticker in the log (OpenSIPS 2.1)?

I was just speaking with bogdan, apparently i am wrong.  he said:

"no, there is no way to do that, as right now any process (SIP 
worker) can execute (if free) the timer jobs, and the 
timer_partitions in TM has nothing to do with that change (in how 
timer jobs are executed)"


So ... i guess you need to prevent any type of blocking operations ...

-Eric

On 02/23/2016 01:34 PM, Rodrigo Pimenta Carvalho wrote:


Ok Eric.

Thank you. I will try  this and see what will be the result.

Thanks a lot.


RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979



*De:* users-boun...@lists.opensips.org 
<users-boun...@lists.opensips.org> em nome de Eric Tamme 
<e...@uphreak.com>

*Enviado:* terça-feira, 23 de fevereiro de 2016 17:23
*Para:* OpenSIPS users mailling list
*Assunto:* Re: [OpenSIPS-Users] What does exactally mean lots of 
warnings about timer_ticker in the log (OpenSIPS 2.1)?
I believe this has been refactored under 2.X as a result ofthe async 
work.  I think the new settings are "timer partitions" here is the 
tm documentation discussing the timer partition setting: 
http://www.opensips.org/html/docs/modules/2.1.x/tm.html#id294483
tm Module - OpenSIPS 
<http://www.opensips.org/html/docs/modules/2.1.x/tm.html#id294483>

www.opensips.org
TM module enables stateful processing of SIP transactions. The main 
use of stateful logic, which is costly in terms of memory and CPU, 
is some services inherently ...




-Eric

On 02/23/2016 12:57 PM, Rodrigo Pimenta Carvalho wrote:


Hi Eric Tamme.


I have just searched about Timer in the module docs, but I didn't 
find any thing about how to manager Timers in OpenSIPS, as you 
commented.


Do you know what part of the documentation tells about "dedicated 
timer processes"?



BTW, my project is embedded.


Any hint will be very helpful!


Thanks a lot!


RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979



*De:* users-boun...@lists.opensips.org 
<users-boun...@lists.opensips.org> em nome de Eric Tamme 
<e...@uphreak.com>

*Enviado:* segunda-feira, 22 de fevereiro de 2016 14:35
*Para:* OpenSIPS users mailling list
*Assunto:* Re: [OpenSIPS-Users] What does exactally mean lots of 
warnings about timer_ticker in the log (OpenSIPS 2.1)?
Many thing

Re: [OpenSIPS-Users] What does exactally mean lots of warnings about timer_ticker in the log (OpenSIPS 2.1)?

[Eric Tamme]

Hey again Rodrigo,

I had further discussions with Bogdan which I will summarize here in an 
attempt to clarify.


Because the new 2.x system is implemented as a reactor, the 
implementation of timer based callbacks has changed.  There is a single 
timer process who's only job is to keep track of jobs that are to be 
scheduled, when it decides a job/callback needs to happen because of the 
current time, it writes to a shared pipe which is read by ALL other 
processes of the reactor.  Any reactor process that is free, will take 
this callback off the pipe and run it.  In this way, there is no way to 
actually block the timer process.


However, what can happen, and what I believe you are seeing here is that 
the timer process has not finished going through the list of timer 
handlers and it should have started processing the list again already - 
so the timer list scan is overlapping.


I'm not 100% sure how this can happen, perhaps you have a very long 
timer list and it is causing the timer process to fail to complete 
before its next scheduled iteration - either than or it is not able to 
write to the shared pipe for some reason.


I would bring the question to either IRC, or to the devel list with as 
much detail as possible.


-Eric


On 02/23/2016 01:59 PM, Rodrigo Pimenta Carvalho wrote:


Ok Eric.


Thank you very much.

I will analyze my OpenSIPS configuration and try to discover what 
action is causing the block situation.



Best regards.


RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979



*De:* users-boun...@lists.opensips.org 
<users-boun...@lists.opensips.org> em nome de Eric Tamme 
<e...@uphreak.com>

*Enviado:* terça-feira, 23 de fevereiro de 2016 17:51
*Para:* OpenSIPS users mailling list
*Assunto:* Re: [OpenSIPS-Users] What does exactally mean lots of 
warnings about timer_ticker in the log (OpenSIPS 2.1)?

I was just speaking with bogdan, apparently i am wrong. he said:

"no, there is no way to do that, as right now any process (SIP worker) 
can execute (if free) the timer jobs, and the timer_partitions in TM 
has nothing to do with that change (in how timer jobs are executed)"


So ... i guess you need to prevent any type of blocking operations ...

-Eric

On 02/23/2016 01:34 PM, Rodrigo Pimenta Carvalho wrote:


Ok Eric.

Thank you. I will try  this and see what will be the result.

Thanks a lot.


RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979



*De:* users-boun...@lists.opensips.org 
<users-boun...@lists.opensips.org> em nome de Eric Tamme 
<e...@uphreak.com>

*Enviado:* terça-feira, 23 de fevereiro de 2016 17:23
*Para:* OpenSIPS users mailling list
*Assunto:* Re: [OpenSIPS-Users] What does exactally mean lots of 
warnings about timer_ticker in the log (OpenSIPS 2.1)?
I believe this has been refactored under 2.X as a result ofthe async 
work.  I think the new settings are "timer partitions" here is the tm 
documentation discussing the timer partition setting: 
http://www.opensips.org/html/docs/modules/2.1.x/tm.html#id294483
tm Module - OpenSIPS 
<http://www.opensips.org/html/docs/modules/2.1.x/tm.html#id294483>

www.opensips.org
TM module enables stateful processing of SIP transactions. The main 
use of stateful logic, which is costly in terms of memory and CPU, is 
some services inherently ...




-Eric

On 02/23/2016 12:57 PM, Rodrigo Pimenta Carvalho wrote:


Hi Eric Tamme.


I have just searched about Timer in the module docs, but I didn't 
find any thing about how to manager Timers in OpenSIPS, as you 
commented.


Do you know what part of the documentation tells about "dedicated 
timer processes"?



BTW, my project is embedded.


Any hint will be very helpful!


Thanks a lot!


RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979



*De:* users-boun...@lists.opensips.org 
<users-boun...@lists.opensips.org> em nome de Eric Tamme 
<e...@uphreak.com>

*Enviado:* segunda-feira, 22 de fevereiro de 2016 14:35
*Para:* OpenSIPS users mailling list
*Assunto:* Re: [OpenSIPS-Users] What does exactally mean lots of 
warnings about timer_ticker in the log (OpenSIPS 2.1)?
Many things share the same timer, dialog expiration cleanup, 
userlocation cleanup, nathelper pings etc.  If one of the processes 
that is triggered by a timer route blocks for some period of time - 
aka a hung db query, it will cause the timer to "drift" and you will 
get logs indicating that a process that should have takeng X amount 
of time actually took Y amount of time - this is due to the blocking 
operation.


I would suggest that you use dedicated timer processes for as many 
things as you can - dia

Re: [OpenSIPS-Users] What does exactally mean lots of warnings about timer_ticker in the log (OpenSIPS 2.1)?

[Eric Tamme]

I was just speaking with bogdan, apparently i am wrong.  he said:

"no, there is no way to do that, as right now any process (SIP worker) 
can execute (if free) the timer jobs, and the timer_partitions in TM has 
nothing to do with that change (in how timer jobs are executed)"


So ... i guess you need to prevent any type of blocking operations ...

-Eric

On 02/23/2016 01:34 PM, Rodrigo Pimenta Carvalho wrote:


Ok Eric.

Thank you. I will try  this and see what will be the result.

Thanks a lot.


RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979



*De:* users-boun...@lists.opensips.org 
<users-boun...@lists.opensips.org> em nome de Eric Tamme 
<e...@uphreak.com>

*Enviado:* terça-feira, 23 de fevereiro de 2016 17:23
*Para:* OpenSIPS users mailling list
*Assunto:* Re: [OpenSIPS-Users] What does exactally mean lots of 
warnings about timer_ticker in the log (OpenSIPS 2.1)?
I believe this has been refactored under 2.X as a result ofthe async 
work.  I think the new settings are "timer partitions" here is the tm 
documentation discussing the timer partition setting: 
http://www.opensips.org/html/docs/modules/2.1.x/tm.html#id294483
tm Module - OpenSIPS 
<http://www.opensips.org/html/docs/modules/2.1.x/tm.html#id294483>

www.opensips.org
TM module enables stateful processing of SIP transactions. The main 
use of stateful logic, which is costly in terms of memory and CPU, is 
some services inherently ...




-Eric

On 02/23/2016 12:57 PM, Rodrigo Pimenta Carvalho wrote:


Hi Eric Tamme.


I have just searched about Timer in the module docs, but I didn't 
find any thing about how to manager Timers in OpenSIPS, as you commented.


Do you know what part of the documentation tells about "dedicated 
timer processes"?



BTW, my project is embedded.


Any hint will be very helpful!


Thanks a lot!


RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979



*De:* users-boun...@lists.opensips.org 
<users-boun...@lists.opensips.org> em nome de Eric Tamme 
<e...@uphreak.com>

*Enviado:* segunda-feira, 22 de fevereiro de 2016 14:35
*Para:* OpenSIPS users mailling list
*Assunto:* Re: [OpenSIPS-Users] What does exactally mean lots of 
warnings about timer_ticker in the log (OpenSIPS 2.1)?
Many things share the same timer, dialog expiration cleanup, 
userlocation cleanup, nathelper pings etc.  If one of the processes 
that is triggered by a timer route blocks for some period of time - 
aka a hung db query, it will cause the timer to "drift" and you will 
get logs indicating that a process that should have takeng X amount 
of time actually took Y amount of time - this is due to the blocking 
operation.


I would suggest that you use dedicated timer processes for as many 
things as you can - dialog, tm etc. see the module docs for how to do 
this.


-Eric

On 02/22/2016 10:28 AM, Rodrigo Pimenta Carvalho wrote:



Hi.

The log of my OpenSIPS began to present thousands of warnings 
similar to:




"WARNING:core:timer_ticker: timer task  already schedualed 
for 93991120 ms (now 248157560 ms), it may ove rlap.."



What does exactly means it?

Some discussions from the past said something about increasing the 
number in the global variable "children".



Should I change the number for children? Before trying it i would 
like to hear something about, just to know better what I'm going to do.



Any hint will be very helpful!


Thanks a lot.



RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979


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Re: [OpenSIPS-Users] What does exactally mean lots of warnings about timer_ticker in the log (OpenSIPS 2.1)?

[Eric Tamme]

I believe this has been refactored under 2.X as a result ofthe async 
work.  I think the new settings are "timer partitions" here is the tm 
documentation discussing the timer partition setting: 
http://www.opensips.org/html/docs/modules/2.1.x/tm.html#id294483


-Eric

On 02/23/2016 12:57 PM, Rodrigo Pimenta Carvalho wrote:


Hi Eric Tamme.


I have just searched about Timer in the module docs, but I didn't find 
any thing about how to manager Timers in OpenSIPS, as you commented.


Do you know what part of the documentation tells about "dedicated 
timer processes"?



BTW, my project is embedded.


Any hint will be very helpful!


Thanks a lot!


RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979



*De:* users-boun...@lists.opensips.org 
<users-boun...@lists.opensips.org> em nome de Eric Tamme 
<e...@uphreak.com>

*Enviado:* segunda-feira, 22 de fevereiro de 2016 14:35
*Para:* OpenSIPS users mailling list
*Assunto:* Re: [OpenSIPS-Users] What does exactally mean lots of 
warnings about timer_ticker in the log (OpenSIPS 2.1)?
Many things share the same timer, dialog expiration cleanup, 
userlocation cleanup, nathelper pings etc.  If one of the processes 
that is triggered by a timer route blocks for some period of time - 
aka a hung db query, it will cause the timer to "drift" and you will 
get logs indicating that a process that should have takeng X amount of 
time actually took Y amount of time - this is due to the blocking 
operation.


I would suggest that you use dedicated timer processes for as many 
things as you can - dialog, tm etc. see the module docs for how to do 
this.


-Eric

On 02/22/2016 10:28 AM, Rodrigo Pimenta Carvalho wrote:



Hi.

The log of my OpenSIPS began to present thousands of warnings similar to:



"WARNING:core:timer_ticker: timer task  already schedualed 
for 93991120 ms (now 248157560 ms), it may ove rlap.."



What does exactly means it?

Some discussions from the past said something about increasing the 
number in the global variable "children".



Should I change the number for children?  Before trying it i would 
like to hear something about, just to know better what I'm going to do.



Any hint will be very helpful!


Thanks a lot.



RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979


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Re: [OpenSIPS-Users] What does exactally mean lots of warnings about timer_ticker in the log (OpenSIPS 2.1)?

[Eric Tamme]

Many things share the same timer, dialog expiration cleanup, 
userlocation cleanup, nathelper pings etc.  If one of the processes that 
is triggered by a timer route blocks for some period of time - aka a 
hung db query, it will cause the timer to "drift" and you will get logs 
indicating that a process that should have takeng X amount of time 
actually took Y amount of time - this is due to the blocking operation.


I would suggest that you use dedicated timer processes for as many 
things as you can - dialog, tm etc. see the module docs for how to do this.


-Eric

On 02/22/2016 10:28 AM, Rodrigo Pimenta Carvalho wrote:



Hi.

The log of my OpenSIPS began to present thousands of warnings similar to:



"WARNING:core:timer_ticker: timer task  already schedualed 
for 93991120 ms (now 248157560 ms), it may ove rlap.."



What does exactly means it?

Some discussions from the past said something about increasing the 
number in the global variable "children".



Should I change the number for children?  Before trying it i would 
like to hear something about, just to know better what I'm going to do.



Any hint will be very helpful!


Thanks a lot.



RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979


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Re: [OpenSIPS-Users] Engage rtpengine if destination resigstered client is ipv6

[Eric Tamme]

The proper way to do this is to check when the user registers if they 
are ipv4 or ipv6 and set a branch flag that will be saved with the 
registration.  When you call lookup, it will restore the branch flag, 
allowing you access to the information about the UAS ipv4, ipv6, behind 
a not or not etc.


-Eric

On 02/08/2016 09:52 AM, Jonathan Hunter wrote:

Hi Eric,


 
Thanks for the response on this.

The main issue was with identifying if the
   Registered user I was sending the INVITE request to
   was over IPv6 or IPv4, and it looks like your config
   should give me an indication.
Many thanks


 
Jon



 
Message: 5


   Date: Mon, 8 Feb 2016 07:44:18 -0700

   From: Eric Tamme <e...@uphreak.com>

   Subject: Re: [OpenSIPS-Users] Engage rtpengine if
   destination

   resigstered client is ipv6

   To: OpenSIPS users mailling list
   <users@lists.opensips.org>

   Message-ID: <56b8a9c2.6090...@uphreak.com>

   Content-Type: text/plain; charset="windows-1252";
   Format="flowed"

   


   You can see an example of setting the flags in my
   federated-sip project

   
https://github.com/etamme/federated-sip/blob/master/core/opensips.cfg.erb#L558

   


   -Eric


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Re: [OpenSIPS-Users] Engage rtpengine if destination resigstered client is ipv6

[Eric Tamme]

yes it is possible to bridge ip4 to ip6 using the "address family" key 
in the offer/answer to specify ipv4 or ipv6.  See the RTPEngine docs for 
more info https://github.com/sipwise/rtpengine/


-Eric

On 02/08/2016 07:38 AM, Jonathan Hunter wrote:

Hi All,

Doe's anyone know if its possible to set for examaple rtpengine to 
bridge between IPv4 and IPv6 only if the target request is destined 
for a subscriber who is registered over IPv6?


I am aware of the address family function, to  determine which 
interface/IP version a SIP request comes from, but is there a similar 
one for destination after a location lookup has been completed?


I am setting a flag when an IPv6 user registers, and was thinking I 
could possibly use an onsend route but just wondered if there is a 
better/easier solution?


This is in an environment with IPv4 and IPv6 users and equipment so I 
only want to engage rtpengine when I need to bridge between IPv4 and v6.


Thanks

Jon


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Re: [OpenSIPS-Users] Engage rtpengine if destination resigstered client is ipv6

[Eric Tamme]

You can see an example of setting the flags in my federated-sip project 
https://github.com/etamme/federated-sip/blob/master/core/opensips.cfg.erb#L558


-Eric

On 02/08/2016 07:38 AM, Jonathan Hunter wrote:

Hi All,

Doe's anyone know if its possible to set for examaple rtpengine to 
bridge between IPv4 and IPv6 only if the target request is destined 
for a subscriber who is registered over IPv6?


I am aware of the address family function, to  determine which 
interface/IP version a SIP request comes from, but is there a similar 
one for destination after a location lookup has been completed?


I am setting a flag when an IPv6 user registers, and was thinking I 
could possibly use an onsend route but just wondered if there is a 
better/easier solution?


This is in an environment with IPv4 and IPv6 users and equipment so I 
only want to engage rtpengine when I need to bridge between IPv4 and v6.


Thanks

Jon


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Re: [OpenSIPS-Users] Error on idle WSS

[Eric Tamme]

This basically happens once the websocket has closed, but opensips still 
thinks there is an active client.  it tries to write some data to the 
socket and it fails.  Its not a "error" really .. but it does make the 
logs dirty.


-Eric

On 01/28/2016 02:59 PM, Sebastian Sastre wrote:

Hey guys,

i’m seeing something in the logs im not quite sure how to interpret. 
Im using sip.js and im able to register and make calls without a problem.


But on the logs, even when i only have one extension registered. i see 
this log randomly.


Jan 28 16:58:37 Proxy01 /sbin/opensips[3394]: 
ERROR:proto_wss:ls_write: TLS connection to xxx.xxx.xxx.xxx:53257 
write failed
Jan 28 16:58:37 CallCenterProxy01 /sbin/opensips[3394]: 
ERROR:proto_wss:tls_write: TLS write error:
Jan 28 16:58:37 Proxy01 /sbin/opensips[3394]: 
ERROR:proto_wss:ls_blocking_write: TLS failed to send data


i can still make calls and everything seems to flow fine. Not sure 
where the error comes from…



any ideas?



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Re: [OpenSIPS-Users] How to take control of Record-Routes header fields? Need remove one of two.

[Eric Tamme]

Perhaps you should try to find out why there are now more RR headers - 
it is likely that the request is spiraling through the proxy itself.  
Generally speaking you should not remove the record route headers.


-Eric

On 01/14/2016 02:16 PM, Rodrigo Pimenta Carvalho wrote:


Dear OpenSIPS-Users;


Suddenly my opensips started to add 2 Record-Routes header fields to 
messages it is handling. Ex: when forward a simple SIP-INVITE.



One header field has a private IP and another has a public IP.


I would like to take control of Record-Routes and then avoid opensips 
adding that one with private IP.



Is it possible?


If yes, when a SIP-OK arrives in a callee, the callee will send 
SIP-ACK to the public IP and then the OpenSIPS will receive it. If I 
let the header field with private IP there, it causes problem when 
opensips is behind a NAT, with callee, but the caller is outside the 
local network.



Can I proceed removing one Record-Route or it is a bad idea?


Any hint will be very helpful!


Best regards.


RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979


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Re: [OpenSIPS-Users] opensips wss support

[Eric Tamme]

For wss to work, your site must be https, and your sip server must be 
wss.  You can not downgrade the connection anywhere.  Also if you are 
linking to sip.js, the url MUST ALSO BE https e.g:


src="https://cdn.rawgit.com/onsip/SIP.js/0.7.2/dist/sip-0.7.2.min.js 
https://cdn.rawgit.com/onsip/SIP.js/0.7.2/dist/sip-0.7.2.min.js>">



For 2.X you must use the new tls_mgm module as the cert management is 
now shared between tls and wss.


for example:

listen=wss:123.456.789.987:443
listen=tls:123.456.789.987:5061

loadmodule "tls_mgm.so"
loadmodule "proto_tls.so"
loadmodule "proto_wss.so"


modparam("tls_mgm", "certificate","/etc/letsencrypt/live/acme.com/cert.pem")
modparam("tls_mgm", 
"private_key","/etc/letsencrypt/live/acme.com/privkey.pem")



I am using sip.js 0.7.2 with latest master from opensips and doing full 
https + wss


-Eric


On 01/12/2016 02:54 PM, Tito Cumpen wrote:

Do Nguyen Ha,

I was getting errors about the ws destination being unencrypted when 
using sip.js.



Razvan or community,

What cert configuration does it utilize? does it take it from the tls 
configuration?


http://www.opensips.org/Documentation/Tutorials-TLS-2-1

In this tutorial I see the protocol being defined. Would these 
settings apply or would a respective proto_wss have to be provided in 
the config?



Thanks,
Tito

On Tue, Jan 5, 2016 at 7:17 AM, Do Nguyen Ha > wrote:


hi Tito

i did test the webrtc with chrome version 47. we only need to
setup web server with HTTPS only that point to jssip source code.
the jssip should works properly

there is no need wss on opensips server - just use ws on opensips

thank you

On Jan 5, 2016 5:53 PM, "Tito Cumpen" > wrote:

Hey Razvan,

Any updates on wss?

On Dec 7, 2015 12:35 PM, "Tito Cumpen" > wrote:

Hey Răzvan,

Any updates on this? Getuser media has been disabled when
using http and using https enforces wss on chrome.

Thanks,
Tito


On Fri, Nov 20, 2015 at 4:44 AM, Răzvan Crainea
> wrote:

Hi, Tito!

I am working on it. I have already started a local
branch and started coding, hopefully it will be public
by the end of the month.

Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com


On 11/19/2015 07:35 PM, Tito Cumpen wrote:

Razvan,


Is there currently a timeline for wss as of now? I
can aid in testing if needed.




On Wed, Nov 4, 2015 at 1:25 PM, Tito Cumpen
> wrote:

Razvan,


Thanks for the reply. Can't wait to try this
feature out.

On Tue, Nov 3, 2015 at 3:32 AM, Răzvan Crainea
> wrote:

Hi, Tito!

Yes, we have WSS support on top of our
priorities, and most likely will be released
with the next OpenSIPS 2.2.

Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com


On 10/16/2015 01:41 AM, Tito Cumpen wrote:

To opensips devs.


Are there any plans to support wss?
Currently chrome will not allow for an
unsecure websocket to be initiated when
using https. They also threaten to remove
getusermedia when using ws.


https://sites.google.com/a/chromium.org/dev/Home/chromium-security/deprecating-powerful-features-on-insecure-origins

Please advise. I am working on a project
that depends on this.

Thanks,
Tito


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