Hello
I am trying to make a test setup to test the RTPENGINE.
Case 1:
Now my setup is Kazoo . i.e.:
Subsriber A ---Asterisk (192.168.175.136)-----via NAT(10.111.101.3)-------------Kamailio(10.10.11.104) ----FreeSwitch(10.10.11.104)
Subscriber B is connected directly to Kamailio (this subscriber is in the same network)
Case 2:
When A calls to B - the sound both ways, so it is ok.
Subsriber A ---Asterisk (10.20.20.101)-----via NAT(10.11.97.12)-------------Kamailio(10.10.11.104) ----FreeSwitch(10.10.11.104)
Subscriber B is connected directly to Kamailio (this subscriber is in the same network)
WITH RTPENGINE:
When A calls to B - the sound is absent, no sound.
I compared both dumps the difference only in addressesAnd what I read in nathelper module:
- 1 - Contact header field is searched for occurrence of RFC1918 addresses.
- 2 - the "received" test is used: address in Via is compared against source IP address of signaling
- 4 - Top Most VIA is searched for occurrence of RFC1918 addresses
- 8 - SDP is searched for occurrence of RFC1918 addresses
- 16 - test if the source port is different from the port in Via
- 32 - test if the source IP address of signaling is a RFC1918 address
All flags can be bitwise combined, the test returns true if any of the tests identified a NAT.
in nat-traversal-role.cfgI see: if (nat_uac_test("3")) { force_rport(); fix_nated_contact(); }
if (has_body("application/sdp") && nat_uac_test("8")) { fix_nated_sdp("10"); }
As I understand - both addresses are RFC1918but in case 2 the RTPENGINE is not kernalized
Does the RTPENGINE distinguish the ip address? why it was not kernalized in case 2?
Any help is appresiated.
Hi
I have the Kamailio and FS behind it (in Kazoo). Both are in one ip address 10.34.101.104
i use rtpengine_manage().
When I call from Asterisk which is behind NAT :1) Asterisk in VMware player (NAT, local ip 192.168.175.136) and public 10.34.101.103 - the voice was ok? in logs I saw Kernalizing media
2)Asterisk in KVM(NAT, local 10.28.40.101 public 10.101.29.45) and no Kernalizing in /var/log/messages and no voice.
The Asterisk configs are absolutely the same.
Kamailio log shows no errors
In both cases the INVITE's SDP connection is 10.34.101.104 - the address of the Kamailioin the ' case 1' I see all RTP streams : 2 rtp streams - between asterisk and kamailio and vice versa, 2 -rtp streams between kamailio/rtpengine - Freeswitch and vice versa
in the second case I see the following RTP streams: 1 rtp stream - from Kamailio to Asterisk, 1 rtp stream - from Freeswitch to Kamailio
How to debug such cases?
Any help is appreciated.
Dear All,
I'd like to send a HTTP call to a HTTP server whenever a call routed through a Kamailio server ends. By looking at the document, seems I should use http_query() function in the utils module. I wonder which route function I should put the http_query() in assuming we use the default kamailio.cfg. Thanks.
Regards,Jay
Hi guys
when i try to register i have route[LOCATION] that returns "404 Not Found location" due to this statement:
$avp(oexten) = $rU;
if (!lookup("location")) {
$var(rc) = $rc;
route(TOVOICEMAIL);
t_newtran();
switch ($var(rc)) {
case -1:
case -3:
send_reply("404", "Not Found location");
exit;
case -2:
send_reply("405", "Method Not Allowed");
exit;
}
}
thats weird i did follow ubuntu realtime trying to get sip account registered and then uac_req_send to asterisk but the first step already failed. :-(
i noticed there is no Username in my RURI (whatever UAC i am using) maybe this is the root of evil
please find attache my kamailio.cfg
i am using kamailio v 4.4
Any idea very welcome
Hello,
I need to know what are the available parameters to set ToS on Kamailio.
I couldn't found much information about that,
only this https://www.kamailio.org/wiki/cookbooks/4.0.x/core#tos
but there is no reference for the available parameters.
Thanks,
Marcelo.
hello
i read the documentation about RTPengine.and the documentation says that:
If INVITE with SDP, when the tm module is loaded, mark transaction with internal flag FL_SDP_BODY to know that the 1xx and 2xx are for rtpengine_answer()
what does it mean?
actually - according to 3261 RTP media (both ways) may be turned on after ACK
Could I get some clarification on tos global setting as referenced by documentation below?
Specifically, are DSCP values allowed? Or is this simply for scheduling purposes at the OS level?
I can change the value once packets hit the network but if possible would like 0x60 to originate from Kamailio.
tos
The TOS (Type Of Service) to be used for the sent IP packages (both TCP and UDP).
Example of usage:
tos=IPTOS_LOWDELAY
tos=0x10
tos=IPTOS_RELIABILITY
Thanks!
-Dan
This report was sent to the Free-RTC list
https://lists.fsfe.org/mailman/listinfo/free-rtc
Most of the projects have an emphasis on SIP
If anybody wants to facilitate testing the new Telepathy/Empathy SIP
connection manager code against Kamailio or other SIP proxies,
registrars and presence servers that would be very welcome.
-------- Forwarded Message --------
Subject: [Free-RTC] GSoC progress, more mentors?
Date: Mon, 13 Jun 2016 22:53:46 +0200
From: Daniel Pocock <daniel(a)pocock.pro>
Reply-To: Free real-time-communications discussion list
<free-rtc(a)lists.fsfe.org>
To: Free-RTC(a)lists.fsfe.org
Hi all,
The GSoC students have officially been working for 3 weeks now.
I created a summary of all the students here:
https://wiki.debian.org/SummerOfCode2016/RTCProjects
They've been publishing weekly reports on the debian-outreach list:
https://lists.debian.org/debian-outreach/
It would be really helpful to increase the size of the mentoring team
for the students working on RTC projects:
- we probably need at least one more Python developer and one more C++
developer to help
- other people can also help test the projects, for example, if you are
using Wordpress, please try Mesut's new WPCall module:
https://github.com/mesutcang/wpcall
As a bonus, some mentors are selected every year for a fully paid trip
to California for Google's GSoC mentor summit.
If anybody has more questions about it or would like to be more involved
please contact me privately.
Regards,
Daniel
_______________________________________________
Free-RTC mailing list
Free-RTC(a)lists.fsfe.org
https://lists.fsfe.org/mailman/listinfo/free-rtc
Hello,
I currently have Freeswitch acting as a B2BUA, handling registrations,
routing, etc for a prototype voice application I have built.
Now I'm at the point where it's time to actually scale it out, and I'm
looking at a few different options for the SIP proxy + registration later.
Essentially, what I would like to have (I think), is:
1) Stateless SIP proxy with consistent hashing to route to:
2) Transaction-stateful SIP proxy handling registrations and initial routing
Ideally, the only thing that's stored across the cluster (if possible)
would be registrations, in something like memcache/redis. Then, of course,
each node might have its own internal state of transactions that it's
tracking. I'd prefer the only thing to care about dialog state to be
Freeswitch.
Finally, the decisions about which FS instance to route to would be made by
my application as well, based on runtime rules.
Is this something people here have done before? Are there other suggestions
on how to scale out kamailio to multiple instances.
Thanks in advance,
Colin
I have spent a couple days beating my head against the Kamilio BLF
feature. But cannot seem to get presence signals from one ua to another.
I am testing with a Zoiper softphone and a Yealink hardware phone, both
registering to Kamailio and able to make and receive calls. But getting
absoultely nowhere with BLF. Have gone through all the suggestions in
whitepaper 43, but still no results. Anybody had success with BLF,
please steer me in the ritght direction.
Thanks, Bill
