Re: [SR-Users] SEMS with sbc application

2018-01-27 Thread Mojtaba 
Hello,
Yes. The log is for sems with mo profile.
I  used SEMS as sbc applications in my network. Let me paste my
configurations below:
#sems.conf
  interfaces=intern,extern

  sip_ip_intern=192.168.18.20
  sip_port_intern=5060
  media_ip_intern=192.168.18.20
  rtp_low_port_intern=2000
  rtp_high_port_intern=5000

  sip_ip_extern=94.182.110.10
  sip_port_extern=4080
  media_ip_extern=94.182.110.10
  rtp_low_port_extern=2000
  rtp_high_port_extern=5000
  public_ip_extern=94.182.110.10
  #sig_sock_opts_extern=force_via_address
  tcp_connect_timeout_extern=1000
  tcp_idle_timeout_extern=90
I was forced to disable "sig_sock_opts_extern" option,

#sbc.conf is like this:

profiles=refuse_with_200,register,mo,mt,refuse
active_profile=$M($m=>methodmap),$M($si=>src_ipmap),refuse
regex_maps=src_ipmap,methodmap

#mo profile is like this:

# defaults: transparent
#RURI=$r
#From=$f
#To=$t
#Contact=
#Call-ID
Call-ID=$ci_leg2
## routing
# outbound proxy:
#outbound_proxy=sip:192.168.5.106:5060
outbound_proxy=sip:192.168.18.19:5060

# force outbound proxy (in-dialog requests)?
#force_outbound_proxy=yes
force_outbound_proxy=yes

# destination IP[:port] for outgoing requests
#next_hop=192.168.5.106:5060
next_hop=192.168.18.19:5060

# set RURI to (calculated) next_hop
#patch_ruri_next_hop=yes
# update next_hop from remote destination? (e.g. from SRV)
#next_hop_fixed=yes
# outbound interface to use (interface ID)
outbound_interface=intern

# SIP NAT handling: recommended if dealing with far end NATs
dlg_nat_handling=yes

## RTP relay
# enable RTP relaying (bridging):
enable_rtprelay=yes
# force symmetric RTP (start with passive mode):
rtprelay_force_symmetric_rtp=yes

# RTP interface to use for A leg
aleg_rtprelay_interface=extern

# RTP interface to use for B leg
rtprelay_interface=intern

#mt profile is like this:

#RURI=$r
#From=$f
#To=$t

#Contact=

#Call-ID
Call-ID=$ci_leg2

## routing
# outbound proxy:
#outbound_proxy=sip:192.168.5.106:5060
#outbound_proxy=sip:$H(P-Route)
#outbound_proxy=sip:$H(P-Source-IP):$H(P-Source-Port)

# force outbound proxy (in-dialog requests)?
force_outbound_proxy=yes
# destination IP[:port] for outgoing requests
#next_hop=192.168.5.106:5060
# set RURI to (calculated) next_hop
#patch_ruri_next_hop=yes
# update next_hop from remote destination? (e.g. from SRV)
#next_hop_fixed=yes
# outbound interface to use (interface ID)
outbound_interface=extern

# SIP NAT handling: recommended if dealing with far end NATs
dlg_nat_handling=yes

## RTP relay
# enable RTP relaying (bridging):
enable_rtprelay=yes
# force symmetric RTP (start with passive mode):
rtprelay_force_symmetric_rtp=yes

# RTP interface to use for A leg
aleg_rtprelay_interface=intern

# RTP interface to use for B leg
rtprelay_interface=extern

## filters:
## filters:
header_filter=blacklist
#header_list=P-App-Param,P-App-Name,P-Route,Remote-Party-ID
header_list=P-Route,Remote-Party-ID,P-Source-IP,P-Source-Port

I think it is big challenge if i am dependent to NAT between the uac
and SEMS (SIP ALG) when i changed default port on sbc.
How can i solved this problem in this regards?
Thanks. With Regards.Mojtaba



On Thu, Jan 25, 2018 at 7:31 PM, Stefan Sayer
 wrote:
> Hello,
>
> Mojtaba wrote on 24.01.2018 12:17:
>> Hello,
>> I have a problem today, It's strange for me.
>> Suppose we have this senario:
>> uac1--->SEMS(mo profile)--->Kamailio>SEMS(mt
>> profile)-->uac2
>>
>> In above topology, we have two interfaces(intern,extern) for SEMS, and
>> just used as SBC (sbc application).
>> if i used port=5060 as external port, every things is right and log
>> file is like this:
>>  [#7fed3f9f9700/32820] [run, udp_trsp.cpp:352] DEBUG: vv M [|] u recvd
>> msg via UDP from 89.165.117.125:42411 vv
>> --++--
>> REGISTER sip:kava.shatel.ir;transport=UDP SIP/2.0
>> Via: SIP/2.0/UDP
>> 89.165.117.125:42411;branch=z9hG4bK-d8754z-021bd8b61efc7ac0-1---d8754z-
>> Max-Forwards: 70
>> Contact: > 89.165.117.125:42411;rinstance=79011092e56e1a09;transport=UDP>
>> To: 
>> From: ;tag=82820e1f
>> Call-ID: Y2U4YThiYjEwNTUzMzliZTIwNWZkMDI3MTM4OTZlNWU.
>> CSeq: 2 REGISTER
>> Expires: 3600
>> Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS,
>> INFO, SUBSCRIBE
>> Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
>> User-Agent: Z 3.3.25608 r25552
>> Allow-Events: presence, kpml
>> Content-Length: 0
>>
>> but when i  changed port=4080 for external port, The Via header and
>> contact header are changed to my public ip, like this:
>>
>>  [#7fed3f9f9700/32820] [run, udp_trsp.cpp:352] DEBUG: vv M [|] u recvd
>> msg via UDP from 89.165.117.125:42411 vv
>> --++--
>> REGISTER sip:kava.shatel.ir;transport=UDP SIP/2.0
>> Via: SIP/2.0/UDP
>> 172.1.1.125:42411;branch=z9hG4bK-d8754z-021bd8b61efc7ac0-1---d8754z-
>> Max-Forwards: 70
>> Contact: 
>> 
>> To: 
>> From: ;tag=82820e1f
>> Call-ID: Y2U4YThiYjEwNTUzMzliZTIwNWZkMDI3MTM4OTZlNWU.
>> CSeq: 2 REGISTER
>> Expires: 3600
>> Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFE

Re: [SR-Users] Kamailio with asterisk on private lan

2018-01-27 Thread Tomi Hakkarainen 
Hi Steve,

I have noticed that too thet Debian is used a lot and myself prefer Ubuntu also 
if I’m asked :) The setup I mentioned is on Suse and all have not went without 
problems.
and currently I have no experience from rtpengine or video.
But I am planning to implement WebRTC also so as Sammy mentioned on his reply 
is the rtpengine way to go with WebRTC.
Also have not yet had problems with rtpproxy 

Tomi

> On 26 Jan 2018, at 20.48, Wilkins, Steve  wrote:
> 
> Hi Tomi,
>  
> I have noticed that applications such as rtpproxy seem to have instructions 
> for Debian type systems, however, I have found that some applications, like 
> rtpengine, do not play as well with Centos; have you had this same 
> experience?  Also, if video is involved, I understood that these proxies are 
> of no benefit.  I don’t know this true or not so please correct me if I am 
> mistaken.
>  
> Thank you,
> -Steve
>   <>
> From: sr-users [mailto:sr-users-boun...@lists.kamailio.org 
> ] On Behalf Of Tomi Hakkarainen
> Sent: Friday, January 26, 2018 12:50 PM
> To: Mark Boyce mailto:m...@darkorigins.com>>
> Cc: Kamailio (SER) - Users Mailing List  >
> Subject: Re: [SR-Users] Kamailio with asterisk on private lan
>  
> Hi,
>  
> I have similar setup working.
> I followed these guides : 
> https://saevolgo.blogspot.fi/2013/08/rtpproxy-revisited-kamailio-40.html 
> 
>  
> Maybe you can find those also useful, and if you need more help just let us 
> know…
>  
> I also wonder when one should use rtpengine over rtpproxy ?
>  
> BR,
> Tomi
> 
> 
> On 26 Jan 2018, at 12.47, Daniel-Constantin Mierla  > wrote:
>  
> To clarify and avoid misleading, rtpproxy should be able to do the same as 
> rtpengine for this case
> Both modules allow to set the public IP address, by providing it as the 
> second parameter to rtp relay manage function. Also, the applications 
> themselves have parameters to specify the address to advertise.
> 
> Cheers,
> Daniel
> 
> On 25.01.18 11:00, Mark Boyce wrote:
> Morning Arsen 
>  
> Thanks I’ll take a look at that.  Was using RTPProxy just through force of 
> habit :-)
>  
> Mark
> 
> 
> On 25 Jan 2018, at 09:53, Arsen  > wrote:
>  
> Hi Mark,
>  
> You can solve this by using rtpengine module, it can rewrite SDP offer / 
> answer and replace media addresses with correct IPs.
>  
>  
> https://kamailio.org/docs/modules/5.0.x/modules/rtpengine.html#rtpengine.f.rtpengine_offer
>  
> 
>  
>  
> 
>  
> Arsen Semionov
> www.eurolan.info 
> cell: +442035198881
>  
> On Thu, Jan 25, 2018 at 11:40 AM, Mark Boyce  > wrote:
> Hi all
> 
> I’m trying to create a relatively simple setup with Kamailio dual homed on 
> public/private ip and asterisk on private ip only.  The idea is load balance 
> / fail over asterisk boxes.
> 
> Following the real-time tutorial I have clients registering with Kamailio, 
> Kamailio registering on clients behalf with asterisk as well as invites going 
> through.
> 
> However what I’m seeing is that when an invite occurs asterisk offers media 
> on its private ip, as it would. However this is making its way through 
> Kamailio all the way to the client.
> 
> After a bit of searching all I can find is people trying to get it working 
> and failing, or putting asterisk on public IP.
> 
> So questions - am I doing this completely the wrong way? Should Kamailio 
> alter the media ip of asterisk on the way through or do I need to do that by 
> hand?  Surely someone somewhere has a write up on this already :-)
> 
> Thanks
> Mark
> 
> ___
> Kamailio (SER) - Users Mailing List
> sr-users@lists.kamailio.org 
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users 
> 
>  
> ___
> Kamailio (SER) - Users Mailing List
> sr-users@lists.kamailio.org 
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users 
> 
>  
> Mark
> -- 
> Mark Boyce
> Dark Origins Ltd
> e: m...@darkorigins.com 
> t: 0845 0043 043
> f: 0845 0043 044
>  
> 
> 
> 
> ___
> Kamailio (SER) - Users Mailing List
> sr-users@lists.kamailio.org 
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users 
> 
> 
> 
> -- 
> Daniel-Constantin Mierla
> www.twitter.com/miconda  -- 
> www.linkedin.com/in/

Re: [SR-Users] Kamailio with asterisk on private lan

2018-01-27 Thread David Villasmil 
If you're implementing webrtc, I'd say you're better off down the road
using rtpengine

On Sat, Jan 27, 2018, 21:38 Tomi Hakkarainen  wrote:

> Hi Steve,
>
> I have noticed that too thet Debian is used a lot and myself prefer Ubuntu
> also if I’m asked :) The setup I mentioned is on Suse and all have not went
> without problems.
> and currently I have no experience from rtpengine or video.
> But I am planning to implement WebRTC also so as Sammy mentioned on his
> reply is the rtpengine way to go with WebRTC.
> Also have not yet had problems with rtpproxy
>
> Tomi
>
>
> On 26 Jan 2018, at 20.48, Wilkins, Steve  wrote:
>
> Hi Tomi,
>
> I have noticed that applications such as rtpproxy seem to have
> instructions for Debian type systems, however, I have found that some
> applications, like rtpengine, do not play as well with Centos; have you had
> this same experience?  Also, if video is involved, I understood that these
> proxies are of no benefit.  I don’t know this true or not so please correct
> me if I am mistaken.
>
> Thank you,
> -Steve
>
> *From:* sr-users [mailto:sr-users-boun...@lists.kamailio.org
> ] *On Behalf Of *Tomi Hakkarainen
> *Sent:* Friday, January 26, 2018 12:50 PM
> *To:* Mark Boyce 
> *Cc:* Kamailio (SER) - Users Mailing List 
> *Subject:* Re: [SR-Users] Kamailio with asterisk on private lan
>
> Hi,
>
> I have similar setup working.
> I followed these guides :
> https://saevolgo.blogspot.fi/2013/08/rtpproxy-revisited-kamailio-40.html
>
> Maybe you can find those also useful, and if you need more help just let
> us know…
>
> I also wonder when one should use rtpengine over rtpproxy ?
>
> BR,
> Tomi
>
>
> On 26 Jan 2018, at 12.47, Daniel-Constantin Mierla 
> wrote:
>
> To clarify and avoid misleading, rtpproxy should be able to do the same as
> rtpengine for this case
>
> Both modules allow to set the public IP address, by providing it as the
> second parameter to rtp relay manage function. Also, the applications
> themselves have parameters to specify the address to advertise.
>
> Cheers,
> Daniel
> On 25.01.18 11:00, Mark Boyce wrote:
>
> Morning Arsen
>
> Thanks I’ll take a look at that.  Was using RTPProxy just through force of
> habit :-)
>
> Mark
>
>
> On 25 Jan 2018, at 09:53, Arsen  wrote:
>
> Hi Mark,
>
> You can solve this by using rtpengine module, it can rewrite SDP offer /
> answer and replace media addresses with correct IPs.
>
>
> https://kamailio.org/docs/modules/5.0.x/modules/rtpengine.html#rtpengine.f.rtpengine_offer
>
>
>
>
> Arsen Semionov
> www.eurolan.info
> cell: +442035198881
>
> On Thu, Jan 25, 2018 at 11:40 AM, Mark Boyce  wrote:
>
> Hi all
>
> I’m trying to create a relatively simple setup with Kamailio dual homed on
> public/private ip and asterisk on private ip only.  The idea is load
> balance / fail over asterisk boxes.
>
> Following the real-time tutorial I have clients registering with Kamailio,
> Kamailio registering on clients behalf with asterisk as well as invites
> going through.
>
> However what I’m seeing is that when an invite occurs asterisk offers
> media on its private ip, as it would. However this is making its way
> through Kamailio all the way to the client.
>
> After a bit of searching all I can find is people trying to get it working
> and failing, or putting asterisk on public IP.
>
> So questions - am I doing this completely the wrong way? Should Kamailio
> alter the media ip of asterisk on the way through or do I need to do that
> by hand?  Surely someone somewhere has a write up on this already :-)
>
> Thanks
> Mark
>
> ___
> Kamailio (SER) - Users Mailing List
> sr-users@lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
>
> ___
> Kamailio (SER) - Users Mailing List
> sr-users@lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
>
> Mark
> --
> Mark Boyce
> Dark Origins Ltd
> e: m...@darkorigins.com
> t: 0845 0043 043
> f: 0845 0043 044
>
>
>
>
> ___
>
> Kamailio (SER) - Users Mailing List
>
> sr-users@lists.kamailio.org
>
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
>
>
> --
>
> Daniel-Constantin Mierla
>
> www.twitter.com/miconda -- www.linkedin.com/in/miconda
>
> Kamailio Advanced Training - March 5-7, 2018, Berlin - www.asipto.com
>
> Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com
>
> ___
> Kamailio (SER) - Users Mailing List
> sr-users@lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
>
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> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
>
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> https://lists.kamailio.org/cgi-bin/

[SR-Users] New module – embedded MSRP relay

2018-01-27 Thread kossivi HOLONOU 
Hello,I would like to develop a project that will allow me to:
   
   - voice, video, screen sharing, etc. sessions with content communication via 
RTP
   - end to end presence – this is purely SIP routing
   - SIMPLE-based presence (aka, presence server or presence agent model) via 
presence* and pua* modules — user presence, dialog states notification (aka, 
blinking lamps), resource lists service (including OMA/RCS extensions), user 
location states notification and replication, audio/video conference mixer 
notifications, a.s.o.
   - embedded XCAP server – management of user contact lists, presence 
policies, user agent configuration files, a.s.o. There is also an XCAP client 
extension
   - embedded HTTP server – for admin and user interaction with the service via 
pure HTTP or XMLRPC requests
   - embedded MSRP relay – for relaying and fine controlling of the 
message-based content of SIP sessions
   - IRC-style instant messaging conference via imc module
   - storage of instant messages for offline users and relay to them when they 
become again online via msilo module
I need your help to start. I went through the document a little without 
understanding much. Thanks for your help
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