Hi guys.
I'm replacing a environment where I was using kamailio + freeswitch by
another where I'm adding rtpengine to the mix.
One of the issues I'm having now is when I have a SIP Client behind NAT:
When I send the INVITE from the SIP Client, the SDP is using the private
LAN IP address + the advertised media port, let's admit it's
10.10.10.10:5000 when the call establishes with a public server running
kamailio + rtpengine, the actual RTP arrives from the home router public IP
on a natted port, let's admit it's 100.100.100.100:65100, event though,
RTPENGINE assumes that RTP stream is coming from 100.100.100.100:5000. I'm
using the '--sip-source' with RTPENGINE to make it use the received ip
address instead of the private IP coming in the SDP but I'm not finding
anything to make RTPENGINE adapt an start sending the RTP FLOW to the port
where he starts receiving the RTP flow from, discarding the SDP media port
information.
In my previous setup with freeswitch alone, freeswitch did this
transparently. How do you guys solve this issue in RTPENGINE?
Regards,
--
Nuno Miguel Reis
Departamento de Engenharia Informática
Faculdade de Ciências e Tecnologia
Universidade de Coimbra
Hello,
during the last developers meeting in Dusseldorf we had some discussions
about simplifying the building process and internal dependencies/libraries.
The src/lib/srutils contains now mainly the uuid and json functions,
which are used by very common modules such as registrar/usrloc, tm,
dialog, jsonrpcs, dmq*, ... There is a part related to the time
recurrence matching, used by drouting, tmrec, but the code is not that
significant in size, so I am planning to relocate the lib code to the
core and get rid of dependencies to internal libsrutils to many modules.
Cheers,
Daniel
--
Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio Advanced Training - March 9-11, 2020, Berlin - www.asipto.com
Kamailio World Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com
Hello,
a quick note to the community forums to announce that the registration
for the next Kamailio World Conference (April 27-29, 2020, in Berlin,
Germany) is now open!
More details are available at:
* https://www.kamailioworld.com/k08/registration/
Call for presentations is still open, there is already a group of very
interesting submissions, we aim to publish details about them in the
near future.
Looking forward to meeting many of you in Berlin!
Cheers,
Daniel
--
Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio Advanced Training - March 9-11, 2020, Berlin - www.asipto.com
Kamailio World Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com
Hi,
I am trying to invoke unum lookups from calls to enum_query() ,is_from_user_enum() and enum_pv_query().
However we are not seeing any enum lookups over the wire using wireshark.
In the Kamailio.cfg the following are set:
loadmodule "enum"
modparam("enum", "domain_suffix", "mydnsserver.org.")
modparam("enum", "i_enum_suffix", " mydnsserver.org.")
sample use in our Kamailio.cfg file:
if(!enum_query("mydnsserver.org.")) {
xlog("....);
#do something
} else {
Xlog("....")
#do something else
}
I do see NAPTR lookups being done by the other Kamailio IMS processes initially when it starts up so I know it has connectivity to the DNS server, but none for the invoked enum queries.
We are running Kamailio 5.2.2.
Thanks,
Martin
Hi,
I am on kamailio 5.2.4 and want to use the xhttp module for certain management operations.
The request involves some DB queries . The DB is remote and via ODBC and might have some delay.The documentation on the xhttp module states about latency http://kamailio.org/docs/modules/devel/modules/xhttp.html mentioning that the request is handled in an asynchronous manner.
However, the JSONRPC-S module documentation says that this module implements the support for asynchronous RPC commands only for HTTP and HTTPS transports.
https://www.kamailio.org/docs/modules/devel/modules/jsonrpcs.html#idm104941…
I am confused about the process under whose context the xttp request . Can there be a dedicated process to handle certain management requests which do some custom instructions like certain DB operations or initiate certain http requests ?
Thanks,Rupesh
I'd never had a reason to before, but on a deployment I need http
messages to go out a specific interface. Curl has an --interface option,
but I don't see how to do the same with http_client. Unless I'm missing
something...
Anyone ever do something like this?
--
Fred Posner
fred(a)qxork.com
https://qxork.com
Direct/SMS: +1 (336) 439-3733
Need Fred? Call Fred. 336-HEY-FRED
Matrix: @fred:matrix.lod.com
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Hi kamailios,
i have a creepy situation with v5.2.1 stable Kamilio. After a day or
so, Kamailio stop to process incoming SIP traffic via TCP. The
incoming TCP network packages get TCP-ACK from the OS (Debian 9,
4.18.0-15-generic-Linux) but Kamailio does not show any processing for
the SIP-Traffic incoming via TCP. No logs, nothing. While traffic via
UDP is working just totally fine.
When i look via command "netstat -ntp" is see, that the Recv-Q get
bigger and bigger. e.g.:
Proto Recv-Q Send-Q Local Address Foreign Address State PID/Program
name tcp 4566 0 172.17.217.12:5060 xxx.xxx.xxx.xxx:57252 ESTABLISHED
31347/kamailio
After Kamailio restart, all is working fine again for a day. We have
maybe 10-20 devices online via TCP and low call volume (1-2 call per
minute). The only settings for tcp we have is "tcp_delayed_ack=no"
How to could we debug this situation? Again, no error, no warings in
the log. Just nothing.
Kristijan
Hi all,
I'm a newbie for Kamailio and working on a feature needs end-users' destination IP in the first transferred/forwarded invite message.
Basically, our system/call flow is like:
ClientA -> Kamailio1 -> Kamailio2 -> ClientB
Invite from the clientA:
tls:212.252.34.165:32980
tls:172.21.193.100:5061
INVITE sip:905364167881@testsip.tvoip.turkcell.com.tr:443 SIP/2.0
Via: SIP/2.0/TLS 100.64.13.139:56993;branch=z9hG4bK.GoiTxA4Ee;rport
From: <sip:905368601286@testsip.tvoip.turkcell.com.tr>;tag=WbINy0ypx
To: sip:905364167881@testsip.tvoip.turkcell.com.tr
CSeq: 20 INVITE
...
Needed Ip (Client B's IP 10.252.180.51 for this case) is in the INVITE row of the forwarded message.
Invite to the Kamailio2 :
udp:172.21.193.100:5060
udp:172.21.193.98:5060
INVITE sip:905364167881@10.252.180.51:36945;transport=tls SIP/2.0
Record-Route: <sip:gearbox11.turkcell.tgc:5060;r2=on;lr;did=7f5.f572;nat=yes>
Record-Route: <sip:testsip05.tvoip.turkcell.com.tr:443;transport=tls;r2=on;lr;did=7f5.f572;nat=yes>
Via: SIP/2.0/UDP gearbox11.turkcell.tgc:5060;branch=z9hG4bK67ad.4e8e863f2475ab8d3053af0593548557.1;i=f
Via: SIP/2.0/TLS 100.64.13.139:56993;received=212.252.34.165;branch=z9hG4bK.1SYkXD1Me;rport=32980
From: <sip:905368601286@testsip.tvoip.turkcell.com.tr>;tag=WbINy0ypx
To: sip:905364167881@testsip.tvoip.turkcell.com.tr
CSeq: 21 INVITE
...
When the first invite message receives, the dialog variables or sel(msg...) parsed values belong to this first received message, as far as I understand during the tests. However I need the forwarded invite message's variables. Even after t_relay in RELAY route, it looks the message I have is the first one. So, I cannot reach this later IP in the forwarded invite message. As far as I understand, bear with me here please, this IP manipulation executes in t_relay function and in the C side of Kamailio. So, I cannot find a way to do that in the our native scripts.
Is there any way to get it without diving C code's abyss? Do you have any idea about it? I would appreciate it, if I can get any help or any clue.
Thanks for all...
S. Cumhur Karahan
[http://s.turkcell.com.tr/SiteAssets/Genel/mail-imza.jpg] <http://turkcell.li/ncfYh>
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Hello,
Long time Kam/Ser user, first time poster.
I'm running into a problem with one of our telco providers when we make a
call that ends up being not in service or some other error. In this case
our ACK's are not working and the phone line stays open for a period
of time until something times out on their end.
They claim the issue is that our negative ACK message is dropping one of
the Via headers. This is the only case I can find in our setup where
Kamailio does this. But it does drop the first Via, which is the first hop
in our internal network.
I don't understand why this is a problem for them, and I'm still trying to
get a reasonable explanation out of them. Technically, I don't see why it
would be a problem. This behaviour is not an issue with our other telco
providers. Strangely enough, it is also not an issue for this provider when
we make the calls over their MPLS network (we are switching to the
internet).
My question is, can this behaviour be changed in Kamailio somehow? Is there
a way for it to keep all the Via headers for negative ACK's?
Or, do I just need to poke them harder to fix their issues?
My setup:
Source -> Kamailio -> Firewall (NAT, SIP ALG) -> Telco Provider
I hope I have provided enough information.
Thanks!
Michael
