Hi,
for sake of Vaclav's courage, let me introduce the "SIP Analysis &
Testing tools".
This is a set of very powerful utilities that may help you to do SIP
call flows analysis. It can do SIP dialogs and transactions matching and
analysis in large pcaps, create pretty call flow diagrams, check and
verify SIP message content (headers, parameters,..) and many more....
It's available on:
https://github.com/vkubart/sipat
For more details, please ask Vaclav Kubart (vaclav.kubart(a)iptel.org).
SIP Analysis & Testing tools
============================
This project offers set of tools for testing SIP.
Dependencies (ubuntu packages)
============
libosip2-dev
libpcap-dev
libxml2-dev
xsltproc
cfanal
======
Call flow analyzer able to draw call flows from PCAP and verify
callflows against given definition.
Usage:
cfanal -r captured-traffic.pcap -a 10.38.14.73=uac -a
10.38.14.74:5062=uas -print-cf drawing.cf
cf2svg drawing.cf drawing.svg
svg2png drawing.svg drawing.png
callflow_diagrams
=================
XSL template for painting call flow diagrams.
.....
Thank you Vaclav a lot!
-Vlada
Where are the instructions for this gui
Im trying to setup a sip load balancer and rtp proxy for nating
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Hello,
a surprise part of 10 Years SER Conference was granting *special* awards
to the people attending the event that had a major contribution in the
management and development of SER/Kamailio to date. Most of them are
well known in the community, but not all of them.
Hope you like them, it proved to be a very entertaining moment. Here is
the pdf with the awards:
* http://asipto.com/u/48
The plan is to extend it with the other people in our community as ideas
with award titles pop up.
Cheers,
Daniel
--
Daniel-Constantin Mierla -- http://www.asipto.com
Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kathttp://linkedin.com/in/miconda -- http://twitter.com/miconda
Hello,
first I want to thank to all participants at 10 Years SER event in
Berlin last Friday. We spent great time, at least I did!
We accepted 50 visiting participants to match the maximum capacity of
the conference room, being full booked about 2 weeks before. Just few
people didn't show up and didn't inform us at all in advance, which was
a pity over all since several persons that registered later were not
accepted, thus some of them could have attended. Folks at Fokus working
with SER/Kamailio joined us and we had probably one of the very few
occasions so far to get so dense knowledge of SIP/VoIP in the same
place, making a day of interesting presentations and open discussions.
Accompanied by nice weather in Berlin, the evening at Playa Paradiso on
river Spree offered a great place for continuing the technical and
mufti-cultural debates (I could count attendees from about 15 countries
(including Brazil!)). Many thanks again to our sponsors for free drinks
and food at this event: Sipgate (http://www.sipgate.de), Amooma
(http://www.amooma.de), NG-Voice (http://www.ng-voice.com), Asipto
(http://www.asipto.com), Frafos (http://www.frafos.com) and Fokus
Institute (http://www.fokus.fraunhofer.de).
I will upload the presentations soon and send an update here -- there
were 15 talks, covering from technical concepts to business matters and
open source philosophy. Therefore if you were a speaker and see this
message, please send the pdf slides to me.
Do not forget, you can join the next party for this celebration this
week in Vienna, details at:
* http://sip-router.org/10-years-ser-vienna/
Meanwhile, be sure you start/continue testing the upcoming release
v3.2.0, we have to be sure it is stable to release it and start a new
development cycle in the near future, to be able to present new features
at the 11th celebration :-) . Here is a step by step guide to pull the
code from GIT and install:
*
http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-devel-from-git
Cheers,
Daniel
--
Daniel-Constantin Mierla -- http://www.asipto.com
Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kathttp://linkedin.com/in/miconda -- http://twitter.com/miconda
Hi,
I am experiencing two problem:
First:
I am experiencing that tls read buffer overflow, if i use default value
of tcp_rd_buf_size
If i doubled the size of read buffer and i use
tcp_rd_buf_size=8192
then the problem seems to be solved.
Second:
I am experiencing that TLS connection is dropped by sip proxy very
frequently!
And i think the tcp connection shouldn't dropped! So my guess is that
TLS communication is not restarting the timeout counters!
So in every 2 minutes the sip-router is restarting the TCP connection
with tcp RST.
like this:
Sep 5 11:28:48 hal /usr/sbin/kamailio[8129]: DEBUG: <core>
[tcp_main.c:4304]: tcp_main: entering timer for 0xac537c74
(ticks=446299809, ti
meout=446299809 (0 s), wr_timeout=0 (240541717 s)), write queue: 0 bytes
Sep 5 11:28:48 hal /usr/sbin/kamailio[8129]: DEBUG: <core>
[tcp_main.c:4346]: tcp_main: timeout for 0xac537c74
Sep 5 11:28:48 hal /usr/sbin/kamailio[8129]: DEBUG: <core>
[io_wait.h:617]: DBG: io_watch_del (0x824cf80, 48, -1, 0x10) fd_no=41 called
Sep 5 11:28:48 hal /usr/sbin/kamailio[8129]: DEBUG: tls
[tls_server.c:597]: Closing SSL connection 0xac4a0374
Sep 5 11:28:48 hal /usr/sbin/kamailio[8129]: DEBUG: <core>
[ip_addr.c:247]: tcpconn_new: new tcp connection: 10.10.10.10
Sep 5 11:28:48 hal /usr/sbin/kamailio[8129]: DEBUG: <core>
[tcp_main.c:1081]: tcpconn_new: on port 36142, type 3
Sep 5 11:28:48 hal /usr/sbin/kamailio[8129]: DEBUG: <core>
[tcp_main.c:1382]: tcpconn_add: hashes: 2473:3781:3367, 5
This second problem is not solved and very urgent for me.
Any help suggestion is highly appreciated!
Thanks,
Misi
Hi,
I need a module which could allow me to send traffic to various carriers and
it has to support some important features. So some basic ones:
- possibility to re-route the call in case the original route fails
- peak/offpeak conditions (time-based)
- route traffic according to prefix
I found LCR and carrierroute module, but it does not have peak/offpeak
feature. Correct me if I am wrong..
Hi all,
I'm happy to announce a new developer for the sip-router project:
Sven Knoblich.
Sven is an experienced C/C++ developer in our offices in Karlsruhe and since
about a year part of the team that develops our VoIP backend services.
Past contributions of him were the CDR based accounting extensions in the acc
module and several other bugfixes and extensions for the dialog module. He'll
maintain in the future this new acc and dialog functionality and also other
code contributed from us.
Best regards,
Henning
--
Henning Westerholt - Head of IT Operations Internet Access & Communications
1&1 Internet AG, Brauerstraße 48, 76135 Karlsruhe, Germany
Hello!
What is it right way to pass all rtp traffic through RTPProxy?
How this can be configured in Kamailio?
Stas Bakulin
----------------------------------------------------------------------
ogion(a)kvant-telecom.ru | www.kvant-telecom.ru
Tel: +7 473 233 0330 (128)
We are trying to configure Kamailio (3.1.x) as a "boarder proxy" where it acts as the front for various carrier gateways so that internal UACs and UASs are unaware of the carrier gateways.
Let me try to present a clear picture of our setup.
1. Kamailio has several NICs (physical or vlan). Each on a different subnet. One subnet is internal/has routes for internal. Other subnets are private connections to carriers or a route to public Internet.
2. All of these subnets are non-routable from Internet. In addition , the carrier private connections are not routable internally.
3. Connection to public internet is via a FW/NAT (one-to-one NAT), which maps to one of the interfaces.
4. All internal UAC/UAS connect on the internal subnet.
5. We are using RTPProxy (at least one instance per carrier) to relay media between internal and carrier subnets
We are able to make this setup up work great except for one scenario. One of the carriers is only reachable via public Internet. Due to security requirements, our Kamailio cannot have a public IP address and must use FW/NAT. I guess this scenario is "Proxy behind NAT" and not really encouraged. But I would like to see if there is a way to make this work. We cannot use the "advertised_address" since it changes the IP for every "route".
We were able to get this mostly working by doing the following
1. mhomed=1
2. Small patch in the rtpproxy module so that force_rtp_proxy actually uses the IP address passed (public IP).
3. Using request_route_preset("publicIP")
The above "mostly" works. By that I mean, the INVITE transaction is properly passed between internal UAS and carrier SBC and the call is setup. However, further transactions (BYE/re-INVITE) etc do not work properly. So, far-end hangups are not working etc.
I've searched various archives of this and other SER lists looking to see if anyone was able to get this scenario working, but couldn't find a definitive answer. Most of them point to using "advertised_address".
So, and ideas on how to make "Proxy behind NAT" work without using advertised_address? Am I wasting my time?
Thanks in advance for any help you can offer.
SV.
