Hi All,
I am using Kamailio IMS VMimage which is provided by Franz. I am
successfully established the call between the IMS clients.
Here I would like to measure the MoS mean opnion score (QoS) of Calls
(Voice quality). Can any one suggest me / guide me how to measure the same
in this IMS network.
Thank you so much in advance.
Regards,
-kranti
Hello,
In a standard sip flow, the call goes like: sip user A --> kamailio --> pstn --> landline user B. However, when user A has a bad internet access, the audio is broken. So what I want is to let sip user A send a invite to kamailio first, then kamailio send invite to user A and B's landline number through pstn, then bridge the two call together.
I understand this can be achieved by using FREESWITCH originate and bridge command. I've tried but there's no audio both ways, which really makes me feel stupid of myself. So I'm wondering if this can be done with kamailio? If so, how?
Thanks
Jesse
Hello,
I've got a bizarre problem caused by bad UA behaviour:
UA A ---> Kamailio (P) ---> UA B
1. UA A sends initial INVITE through P to B;
2. Kamailio (P) makes some modifications to the From header using
uac_replace_from() and passes along to B.
3. B sends an in-dialog request (e.g. BYE or reinvite) to A through P;
in doing so, it modifies the To (formerly From) value slightly,
replacing the hostname portion in the To URI with a different value to
the one that was received in the From header.
4. Kamailio relays this in-dialog request to A, but with an
adulterated/clipped/truncated/grammatically invalid To header now.
5. A responds with 400 Bad Request due to invalid To header.
I would not dispute that UA B should not be modifying the remote URI in
this manner. But since it does, it gives rise to two questions:
1. Why does Kamailio not simply discard the modified To header and
restore the original value stored in the Record-Route rider parameter?
Is it because the Record-Route parameter does not contain the original
header value, but rather some data complementary to the current header
value?
Or is it because the UAC code takes a checksum of the original remote
URI header value and stores it, and just checks it when restoring on
principle? If so, what's the motive for that?
2. Wouldn't it be better behaviour to simply reject a request so
malformed, rather than passing it on with a corrupt restored value? If
Kamailio can detect that the header has been tampered with, why pass it
on?
-- Alex
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
Hello ,
I’m trying to build a testbed using the VMware image you kindly have shared
here: https://www.kamailio.org/w/2016/02/kamailio-ims-getting-started-box/
My configuration details:
HSS - 10.0.0.9 (./hss.sh)
PCSCF - 10.0.0.10 (kamailio -f /usr/local/etc/kamailio/kamailio-pcscf.cfg)
ICSCF - 10.0.0.11(kamailio -f /usr/local/etc/kamailio/kamailio-icscf.cfg)
SCSCF - 10.0.0.12 (kamailio -f /usr/local/etc/kamailio/kamailio-scscf.cfg)
host machine: 10.0.0.5
I am able to register mobile clients ( IMSdroid) and make voice/video calls
between them successfully. Now I would like to make a conference call among
3 IMSdroid users but its failing.
Could you please suggest me the way forward here.
I am thinking to integrate Application server (AS) to enable Video
Conferences. Could you please guide me how to enable AS in this VM image
and configuration details.
Is there any possibility to do Video conference using this current setup(VM
image). Could you please suggest me.
Thank you so much in advance.
Regards,
-kranti
Hi,
I am newbie to kamailio world and we would like to deploy Kamailio based
IMS platform. Can any body share the link or Step by step procedure(user
Guide) to deploy Kamailio IMS.
Regards,
Ramya
hi all;i want to write "billing" project with kamailio,i using python,postgresql, dialog module and falcon.my problem is in "call_id" section.my code(this section) is in python :
"""...
if (method == 'INVITE'):
callid = req.get_param_as_list("call_id")
print(req.get_param_as_list("Ts"))
print("call id = %s " % callid)
elif (method == 'ACK'):
time_req = int(req.get_param_as_list("Ts")[0])
callid = req.get_param_as_list("call_id")[0]
print("user %s call to %s at time %d % (from_user,to_user,time_req))...
"""
and in kamailio.cfg:
"""http_query("http://127.0.0.1:5000/test?method=$rm& call_id=$dlg(callid)&Ts=$Ts&fu=$fu&tu=$tu","$var(result)");
""""
"callid" in ACK is ok but "callid' in invite is "null", where is my problem?
thanks.
Hi all,
I wonder purpose of this block in default script. What I didn’t understand is the purpose of "t_check_trans()”. Accoording to documentation, this method returns a boolean. So why do we use it even we don’t check output?
# handle retransmissions
if (!is_method("ACK")) {
if(t_precheck_trans()) {
t_check_trans();
exit;
}
t_check_trans();
}
/Volkan
I would like to support VoLTE using an IPv4 IMS network rather than IPv6
network. The handsets that I have been trying insist on the IMS network
support either IPv6 or IPv4v6 dual stack. After giving them IPv4v6
connectivity, I am finding that even though the handsets asked for both IPv4
and IPv6 P-CSCFs, they only try the IPv6 proxy (if only the IPv4 proxy is
provided, they will not try at all).
I am wondering of anyone has worked with any LTE handsets which are willing
to support IMS using IPv4 only.
Thanks,
Ron
Dear All,
I am a new entrant in Kamailio though little understanding of SIP and
FreeSwitch soft switch, however, I have a project to integrate Kamailio and
Freeswitch.
Here is the scenario of what I want to implement;
Kamailio will act as our SIP control while Freeswitch will act media server
for incoming calls.
Freeswitch will send all outgoing calls to Kamailio for onward transfer to
our SIP provider network.
No registration, presence, location Accounting, Authentication etc. are
required.
I have gone through the book SIP ROUTING WITH KAMAILIO and FreeSwitch and
Kamailio integration sample config available at
http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms am still
unable to figure out how to go with the configuration.
I will appreciate any one who can provide guideline for achieving the above
scenario.
Regards
Olawuyi Timothy Oladapo | IT/VAS Operation | Information System Dept
|Pethahiah Rehoboth INTL. Limited | 2nd Floor Rubby Block All seasons Plaza
, Lateef Jakande Rd, Agidingbi, Ikeja, Lagos. | Mobile:
+2348052612001,+2348098797928 |Skype: daptims |
<htttp://www.pethahiah.com/> htttp://www.pethahiah.com
Hello,
This is probably trivial, but I'm quite new in kamailio and trying to
handle it for my personal purposes (so - no trainings, no support etc :)
) - for few (up to 20) users and rare connections, more to learn
something than to have production system. No plans to go outside my own
server.
I have LAN. One of machines in this LAN is server with kamailio onboard.
It works, but only with local hostname (user@hostname).
Now I want to go outside (with clients).
As temporarty solution I have only dynamic DNS (no-ip.org) and I didn't
expect it working. But tried with IP adress (user(a)WAN.IP.ADD.RESS).
And... I don't know which ports to redirect on router.
Now there are redirected 5060 and 10000-30000 (maybe it is not necessary
but first I want it working) - taken from some on-line info.
I can:
- log in to all my accounts,
- depends on client - call to another user and in some cases I can
receive the call (so it is probably issue on my clients, because it is
possible to make a call at all).
But I cannot transmit any data: connections are completely silent in any
case and I cannot start video - although cameras are findable.
Which ports in fact are necessary?
I want audio and video communication, presence info and probably nothing
more. Which ports should I redirect?
--
Pozdrawiam
Andrzej Kaczmarczyk
