[SR-Users] sipml5 fail invite kamailio ims web sockets

[Serhat Guler]

Dear all,

I can register the sipml5 client to my kamailio IMS setup successfully, but
when I try to make a call the invite fails with "478 Unresolvable
destination". From the wireshark trace file I think the PCSCF on the
terminating side cannot resolve the destination. After the invite request
fails the caller re-registers to the networkj automatically. The wireshark
file is attached.

The registration process happens successfully but all of the nodes give
some errors:

10(3773) INFO: 

Re: [SR-Users] BYE issue

[Alex Balashov]

Nelson,

You are very correct. The request URI of the BYE (and all other 
in-dialog requests, such as reinvites) should equal the Contact URI of 
the party to which it is being sent, and this should not change even if 
the in-dialog request is being sent through Kamailio because of 
Record-Route.


-- Alex

--
Alex Balashov | Principal | Evariste Systems LLC

Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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Re: [SR-Users] RTPProxy

[Alex Balashov]

On 10/17/2016 02:29 PM, Rodrigo Moreira wrote:


What is difference between modules rtpproxy and rtpengine?


rtpproxy is a userspace process which, historically, has a relatively 
limited call throughput capacity (maybe a few hundred calls), though 
this might be addressed to some degree in rtpproxy 2.0. Nevertheless, it 
has been commonly used and well supported in the *SER family for long time.


RTPEngine is a newer initiative from Sipwise, and uses kernel-mode 
forwarding to achieve close to on-the-wire RTP forwarding speeds. It can 
do 10,000+ concurrent bidirectional RTP streams. It also has lots of 
other features which can be useful in, for example, running an RTP relay 
in 1:1 NAT environments such as AWS, or in enabling WebRTC.


However, it is a bit more complicated to set up than vanilla rtpproxy. 
Not much more, though.


-- Alex

--
Alex Balashov | Principal | Evariste Systems LLC

Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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[SR-Users] RTPProxy

[Rodrigo Moreira]

Hi,

What is difference between modules rtpproxy and rtpengine?

Best regards.
-- 
Rodrigo M.
(37) 9132-4539
(34) 9889-3069
rodrigo.moreira2007
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[SR-Users] BYE issue

[Nelson Migliaro]

Hello everybody,

I am having issues with one SIP vendor.

I have a Kamailio in bridge mode (private IP / Public IP) and some Asterisk
and Media Gateways.

Calls get established and I have two way audio but when the remote party
hangs up the call, the BYE arrives to the Kamailio and does not move
forward.

I think the problem is SIP vendor rewrite the BYE header and change the
asterisk IP with the public IP of the kamailio.

The IP that appears in the header of the BYE have to be the same that
appears in the contact (UAC that send the call, in my case the Asterisk).
Vendor should not change that IP. ¿Am I correct?

Thank you

-
INVITE

2016/10/17 18:50:49.110967 PUBLIC-KAMAILIO-IP:5060 -> VENDOR-IP:6060
INVITE sip:DESTINATION-NUMBER@VENDOR-IP:6060 SIP/2.0
Record-Route:

Record-Route:

Via: SIP/2.0/UDP
PUBLIC-KAMAILIO-IP;branch=z9hG4bK06a.07540d0e2f32a811ecf9c0a5235dc77a.1
Via: SIP/2.0/UDP
PRIVATE-ASTERISK-IP:5060;received=PRIVATE-ASTERISK-IP;branch=z9hG4bK6bb5a7b3;rport=5060
Max-Forwards: 69
From: "SOURCE-NUMBER" ;tag=as5e87b96c
To: 
Contact: 
Call-ID: 025cc3717ba59faa000cf4db6f8be588@PRIVATE-ASTERISK-IP:5060
CSeq: 102 INVITE
User-Agent: UAC
Date: Mon, 17 Oct 2016 16:53:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 269

v=0
o=root 292850421 292850421 IN IP4 PUBLIC-KAMAILIO-IP
s=Asterisk PBX
c=IN IP4 PUBLIC-KAMAILIO-IP
t=0 0
m=audio 23456 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
a=nortpproxy:yes

-
BYE
-
2016/10/17 18:50:58.241666 VENDOR-IP:6060 -> PUBLIC-KAMAILIO-IP:5060
BYE sip:SOURCE-NUMBER@PUBLIC-KAMAILIO-IP:5060 SIP/2.0
Via: SIP/2.0/UDP VENDOR-IP:6060;branch=z9hG4bKeff4.48943e76.0
Via: SIP/2.0/UDP
VENDOR-IP:5060;branch=z9hG4bK1d4e605e4ll19f74fBYE421ce8658050206
Max-Forwards: 34
Route:

Route:

To: "SOURCE-NUMBER";tag=as5e87b96c
From: ;tag=421ce86-co1547-INS001
Call-ID: 025cc3717ba59faa000cf4db6f8be588@PRIVATE-ASTERISK-IP:5060
CSeq: 154701 BYE
User-Agent: VENDOR
Content-Length: 0

-
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Re: [SR-Users] Sip route can be as IMS PCSCF node

[Jason Penton]

Hello,

yes you can to both questions. There are sample configs in the source tree
as well.

Cheers
Jason

On Mon, Oct 17, 2016 at 12:21 PM, Surender Singh  wrote:

> Hi,
>
>
>
> Can we use sip router as IMS PCSCF to connectivity with IMS.
>
>
>
> Does sip router supports the diameter RX interface to connectivity with
> PCRF .
>
>
>
> Regards
>
> Surender Singh
>
>
>
>
>
>
>
> ::DISCLAIMER::
> 
> 
> 
>
> The contents of this e-mail and any attachment(s) are confidential and
> intended for the named recipient(s) only.
> E-mail transmission is not guaranteed to be secure or error-free as
> information could be intercepted, corrupted,
> lost, destroyed, arrive late or incomplete, or may contain viruses in
> transmission. The e mail and its contents
> (with or without referred errors) shall therefore not attach any liability
> on the originator or HCL or its affiliates.
> Views or opinions, if any, presented in this email are solely those of the
> author and may not necessarily reflect the
> views or opinions of HCL or its affiliates. Any form of reproduction,
> dissemination, copying, disclosure, modification,
> distribution and / or publication of this message without the prior
> written consent of authorized representative of
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> please delete it and notify the sender immediately.
> Before opening any email and/or attachments, please check them for viruses
> and other defects.
>
> 
> 
> 
>
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-- 

*Jason Penton*
*Senior Manager: Applications and Services*
*Smile Communications Pty (Ltd)*


*Voice:Mobile:* +234 (0) 702 000 000 7

+27 (0) 83 283 7000
*Skype:* jason.barry.penton
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[SR-Users] Sip route can be as IMS PCSCF node

[Surender Singh]

Hi,

Can we use sip router as IMS PCSCF to connectivity with IMS.

Does sip router supports the diameter RX interface to connectivity with PCRF .

Regards
Surender Singh




::DISCLAIMER::


The contents of this e-mail and any attachment(s) are confidential and intended 
for the named recipient(s) only.
E-mail transmission is not guaranteed to be secure or error-free as information 
could be intercepted, corrupted,
lost, destroyed, arrive late or incomplete, or may contain viruses in 
transmission. The e mail and its contents
(with or without referred errors) shall therefore not attach any liability on 
the originator or HCL or its affiliates.
Views or opinions, if any, presented in this email are solely those of the 
author and may not necessarily reflect the
views or opinions of HCL or its affiliates. Any form of reproduction, 
dissemination, copying, disclosure, modification,
distribution and / or publication of this message without the prior written 
consent of authorized representative of
HCL is strictly prohibited. If you have received this email in error please 
delete it and notify the sender immediately.
Before opening any email and/or attachments, please check them for viruses and 
other defects.


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Re: [SR-Users] sipml5 register to kamailio IMS (Unauthorized - Challenging the UE)

[Serhat Guler]

Hi Daniel,

Thanks for your reply. I dug in a bit more and found out that the sipml5
doesn't calculate the digest correctly when AKAv1MD5 or v2 is selected as
default (have no idea why). So, I set the default authentication algorithm
as Digest MD5 and now I can register the UE.

Cheers,
Serhat

On 17 October 2016 at 11:47, Daniel-Constantin Mierla 
wrote:

> Hello,
>
> maybe you can get more hints about what happens there by running with
> debug=3 inside the kamailio cfg.
>
> Cheers,
> Daniel
>
> On 14/10/16 17:40, Serhat Guler wrote:
>
> Hi,
>
> I am trying to register sipml5 webrtc client to my kamailio IMS setup. I
> have tried to register the client both with ws and wss, but it seems to be
> that the sipml5 doesn't calculate the authentication digest right. The
> authentication mechanism i set to AKAv1-MD5 as default in the hss. A simple
> wireshark file is attached. .10 being the host, .11 being the kamailio
> server.
>
> From the output of scscf we can see that the digests do not match.
>
> [REGISTER] from [sip:b...@net1.test] to [sip:b...@net1.test]
> ims_auth [authorize.c:824]: authenticate(): uri=sip:net1.test
> nonce=rB2iDuerHwoy+LUStSOsYojAESfWmAAApFZ3XNB8FdA= response=
> 61cbdbeb47c9880ededfca51c3801800 qop=auth-int nc=0001 cnonce=
> 2d4545cf4c935c8094c8b1da3d4a2976 hbody=d41d8cd98f00b204e9800998ecf8427e
> ims_auth [authorize.c:872]: authenticate(): UE said:
> 61cbdbeb47c9880ededfca51c3801800 and we expect
> 7e27ef414cf37d96cd1c849bb7e59415 ha1 a53988ba0b257941bc747b1026225c77
> (REGISTER)
> tm [tm.c:1265]: w_t_reply(): ERROR: t_reply: cannot send a t_reply to a
> message for which no T-state has been established
>
> Thanks a lot,
> Serhat
>
>
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>
>
> --
> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - 
> http://www.linkedin.com/in/miconda
> Kamailio Advanced Training, Berlin, Oct 24-26, 2016 - http://www.asipto.com
>
>
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>
>
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[SR-Users] How to use sql transformations?

[Daniel Tryba]

How to use the sql transformation?
https://www.kamailio.org/wiki/cookbooks/4.4.x/transformations#sql_transformations
has the following example:

xlog("$$rm = $rm = $(rm{s.sql})");

But adding this to the request_route and starting kamailio will fail:

ERROR: pv [pv_trans.c:2351]: tr_parse_string(): unknown transformation: 
sql}/sql/3!
ERROR:  [pvapi.c:1629]: tr_lookup(): error parsing [{s.sql}]
ERROR:  [pvapi.c:1010]: pv_parse_spec2(): bad tr in pvar name "rm"
ERROR:  [pvapi.c:1036]: pv_parse_spec2(): invalid parsing in 
[$(rm{s.sql})] at (4)
ERROR: xlog [xlog.c:512]: xdbg_fixup_helper(): wrong format[$$rm = $rm = 
$(rm{s.sql})]
ERROR:  [route.c:1154]: fix_actions(): fixing failed (code=-1) at 
cfg:/etc/kamailio/kamailio.cfg:372

line 372 is the above xlog and sqlops.so is loaded (and works). Anybody 
got a working example of this? Or an other hint to prevent sql
injections when using user supplied variables in sql queries?


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Re: [SR-Users] 503 Service unavailable to 500

[Alex Balashov]

Hello, 

Any reply code/status translation is going to invoke the absorption of the old 
reply and the generation of a new one, within the proxy. Proprietary SIP 
headers would not be conserved in such a scenario. 

To add headers to a reply, you can use append_to_reply(), which works in much 
the same manner as append_hf(). 

-- Alex

--
Principal, Evariste Systems LLC (www.evaristesys.com)

Sent from my Google Nexus.


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[SR-Users] RTPEngine & Homer

[Grant Bagdasarian]

Hello,

I'm trying to get rtpengine to send its stats to Homer, but I'm unsure to which 
IP I should configure RTPengine to send the stats?
We have a Kamailio server configured with the sipcapture module, which works 
perfectly for SIP traffic, but RTP stats are not visible in Homer.
Do I need to point RTPengine to the sip capture server or directly to Homer?

Running RTPEngine 4.5.2 and Kamailio 4.4.1.

Regards,

Grant

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[SR-Users] 503 Service unavailable to 500

[Dhruvin Desai]

Hi

503 Service unavailable received at kamailio SIP server is forwarded as 500
server internal error to the UAC. Which seems fine based on the fact that
there is no retry-after header.

Would like to know if there is any way to to control the SIP Headers like
adding Proprietary SIP Headers to the 500 "Server Interrnal Errror" created
by kamalio stack for 503 service unavailable (without retry-after Header)

It seems that Kamalio does not retain the proprietary sip headers received
in 503  to 500 response forwarded.

Any insight into the issue would be helpful

-- 
Warm Regards

Dhruvin Desai.
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Re: [SR-Users] sipml5 register to kamailio IMS (Unauthorized - Challenging the UE)

[Daniel-Constantin Mierla]

Hello,

maybe you can get more hints about what happens there by running with
debug=3 inside the kamailio cfg.

Cheers,
Daniel


On 14/10/16 17:40, Serhat Guler wrote:
> Hi,
>
> I am trying to register sipml5 webrtc client to my kamailio IMS setup.
> I have tried to register the client both with ws and wss, but it seems
> to be that the sipml5 doesn't calculate the authentication digest
> right. The authentication mechanism i set to AKAv1-MD5 as default in
> the hss. A simple wireshark file is attached. .10 being the host, .11
> being the kamailio server.
>
> From the output of scscf we can see that the digests do not match.
>
> [REGISTER] from [sip:b...@net1.test] to [sip:b...@net1.test]
> ims_auth [authorize.c:824]: authenticate(): uri=sip:net1.test
> nonce=rB2iDuerHwoy+LUStSOsYojAESfWmAAApFZ3XNB8FdA=
> response=61cbdbeb47c9880ededfca51c3801800 qop=auth-int nc=0001
> cnonce=2d4545cf4c935c8094c8b1da3d4a2976
> hbody=d41d8cd98f00b204e9800998ecf8427e
> ims_auth [authorize.c:872]: authenticate(): UE said:
> 61cbdbeb47c9880ededfca51c3801800 and we expect
> 7e27ef414cf37d96cd1c849bb7e59415 ha1 a53988ba0b257941bc747b1026225c77
> (REGISTER)
> tm [tm.c:1265]: w_t_reply(): ERROR: t_reply: cannot send a t_reply to
> a message for which no T-state has been established
>
> Thanks a lot,
> Serhat
>
>
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-- 
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http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Oct 24-26, 2016 - http://www.asipto.com

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Re: [SR-Users] Send SIP Info within a dialog using $uac_req(method)="INFO"

[Daniel-Constantin Mierla]

Hello,

if you want to run an MI command over RPC, you have to use 'mi' as the
rpc command and the MI command as the first parameter, followed by the
rest of the parameters for the command.

Cheers,
Daniel


On 14/10/16 14:23, Jonathan Hunter wrote:
> Hi Daniel,
>
> Also I am trying to fire that command using jsonrpc_exec and I keep
> getting;
>
>
> jsonrpc_exec_ex(): method callback not found [t_uac_dlg]
>
> I have tried with t.uac_dlg and get the same response, can you let me
> know if this command is support with this module on 4.3 please and if
> so what am I doing wrong with the syntax?
>
> Thanks
>
> Jon
>
> 
> From: hunter...@hotmail.com
> To: mico...@gmail.com; sr-users@lists.sip-router.org
> Date: Fri, 14 Oct 2016 08:52:25 +
> Subject: Re: [SR-Users] Send SIP Info within a dialog using
> $uac_req(method)="INFO"
>
> Hi Daniel,
>
> Thanks for the response, sorry I must of missed this!
>
> I was thinking of using the t_uac_dlg command to generate the INFO
> message, but will this allow me to do it within an established INVITE
> dialog?
>
> I am just worried that changing the CSEQ value will cause issues, so
> am I better looking to modify in a B2BUA rather than the proxy, or
> will the dialog module handle this?
>
> Thanks
>
> Jon
>
>
> 
> To: sr-users@lists.sip-router.org
> From: mico...@gmail.com
> Date: Thu, 6 Oct 2016 12:41:32 +0200
> Subject: Re: [SR-Users] Send SIP Info within a dialog using
> $uac_req(method)="INFO"
>
> Hello,
> uac_req_send() is able to send only initial requests (with follow up
> on auth challenge). It doesn't expose the ability to send requests
> within a dialog -- the functions exist in c (tm module), but not
> availble in config.
> On the other hand, there should be a mi/rpc command exported by tm
> module that allows that -- it may be possible to do it from config
> file via jsonrpc-s module.
> Cheers,
> Daniel
>
>
> On 29/09/16 21:41, Jonathan Hunter wrote:
>
> Hi Guys,
>
> Is it still the case that when using uac_req_send, you cant send withing 
> a specific dialog?
>
> I can modify call-id, but I presume tags may be more of a problem?
>
> See old post below from 2015;
>
> >/I am familiar with uac_req_send. but how do I send it with in a 
> />/specific dialog and with data in the INFO req ? /sending a new request 
> inside a dialog is not possible with
> uac_req_send(). It is not easy over all because you change the sequence
> order (CSeq value). Practically, you need to track how many requests you
> sent from the middle to update (and restore in reply) when caller or
> callee sends a new request.
>
> dialog module can track changes in CSeq for requests sent to callee,
> being used now for authentication of INVITE to another provider, when
> Kamailio adds the credentials. But for more you would need to extend the
> dialog module.
>
> I just need to send a SIP info within an established dialog to stop some 
> function up stream, so wondered if this is still a blocker?
>
> Many thanks
>
> Jon
>
>
>
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> -- 
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> http://twitter.com/#!/miconda  - 
> http://www.linkedin.com/in/miconda
> Kamailio Advanced Training, Berlin, Oct 24-26, 2016 - http://www.asipto.com
>
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-- 
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Kamailio Advanced Training, Berlin, Oct 24-26, 2016 - http://www.asipto.com

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Re: [SR-Users] Active calls limit

[Daniel-Constantin Mierla]

Hello,

you can use sqlops module to load data from database and store in an xavp.

If you want to load from subscriber table along with auth credentials,
look at load_credentials parameter from auth_db module.

Cheers,
Daniel


On 14/10/16 12:05, Ivan Dudko wrote:
> Hello!
>
> I am trying to implement limit of active calls for subscribers. And
> for each subscriber this limit must be personal.
>
> I find example route in presentation of Daniel-Constantin Mierla.
> And i need to set $xavp(caller=>active_calls) = 1;
> for each dialog. But i can't understand how to load this number of
> active calls from some-thing similar of user profile or subscriber table?
>
> route[DIALOG] {
>
> if (is_method("CANCEL") || (has_totag() &&
> is_method("INVITE|BYE|ACK"))) {
> dlg_manage();
> return;
> }
>
> if (is_method("INVITE") && !has_totag() &&
> !isflagset(WITH_ACTIVE_CALLS_LIMIT)) {
> if( $xavp(caller[0]=>active_calls) != $null &&
> $xavp(caller[0]=>active_calls) > 0 ) {
> if(!get_profile_size("caller", "$fU@$fd", "$var(acsize)")) {
> send_reply("500", "No more active calls");
> exit;
> }
> if($var(acsize)>=$xavp(caller[0]=>active_calls)) {
> send_reply("403", "No more active calls");
> exit;
> }
> set_dlg_profile("caller", "$fU@$fd");
> }
> setflag(WITH_ACTIVE_CALLS_LIMIT);
> dlg_manage();
> }
> }
>
>
>
>
> ___
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users@lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

-- 
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Oct 24-26, 2016 - http://www.asipto.com

___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users