Hi,
(Kamailio 4.2.3)
Working to get dialogs closed and cleaned up nicely I'm looking into the
ka_interval and ka_timer parameter of the dialog module. The following
is now happening.
Scenario:
1. Call gets setup
2. Callee's internet connection drops
3. Call gets terminated (nicely) due to ka_timer/ka_interval
4. Log fills with "freeing a free fragment" after about 10 seconds
Digging into this I saw that the dialog is being closed and cleaned
properly in step 3. But it fails to remove/clean the timers used for the
keep-alive. That's what triggers step 4, which in turn tries to clean
the (or some of the) dialog space again. Is this expected behaviour when
using the keep-alive options, or am I running into somekind of (known) bug?
Thanks in advance for any help.
--
Cheers,
Dirk Teurlings
Hi All
I am using Kamailio 4.0.4 (x86_64/linux).
I wanted to do simultaneous registrations of large number of endpoints.
Many times I see only 2048 endpoints registered and for other endpoints
Kamailio did not respond for REGISTER message.
I am tcp as transport.
Now my questions are
1. What are the different configurations available to increase Kamailio
performance
2. If I put Kamailio in a Linux 64 Bit, 2GHz, 8GB RAM machine, how much
simultaneous and total registrations can be achieved.
I have tuned system file FDs, tcp related configurations like ephemeral
port range, and other required configurations.
I hope there should be some configurations in Kamailio to achieve higher
throughput , can somebody help me out.
Thanks
Austin
Hi,
I have added the ASTERISK integration in kamailio.cfg, after adding those,
am not able to start kamailio service, its getting failed, attached is the
script.
Any help would be appreciated.
Thank you with regards,
Gopalkrishnan N.
Mob: +91 99404 91346
VoIP call - sip:saigop@gtalk2voip.com
Hello list!
Have some difficulties with RTPengine:
My test scheme is: UA(WS) -> Kamailio (WS proxy) -> Kamailio (SIP REGISTRAR,
proxy) -> ASTERISK (media server) -> UA (SIP)
Kamailio (WS) use RTPengine.
Kamailio (WS proxy) - 4.2.3
Kamailio (SIP REGISTRAR, proxy) - 4.1.4
RTPengine - 3.3.0.0+0~mr3.8.0.0
I make a test call and have troubles in negotiation between UA(WS) ->
Kamailio (WS proxy):
UA(WS) -> INVITE -> Kamailio WS
......
rtpengine_manage("trust-address replace-origin replace-session-connection
ICE=force");
......
this INVITE consists in SDP section a=fingerprint:sha-256......
When Kamailio (WS) receives 200 OK, it is also handled by RTPengine
.......
rtpengine_manage("trust-address replace-origin replace-session-connection
ICE=force");
.......
but UA (WS) receives 200OK without fingerprint, and log an error:
Failed to set remote answer sdp: Called with SDP without DTLS fingerprint.
Currently spend a lot of time reading Kamailio/RTPengine
documentation/mail-list - but without success
If someone have some hints, I would appreciate any help.
--
View this message in context: http://sip-router.1086192.n5.nabble.com/RTPengine-Kamailio-200-OK-without-D…
Sent from the Users mailing list archive at Nabble.com.
when kamailio forwards invite request, it adds to it a new via header.
for example, incoming invite:
SIP/2.0/TCP 192.168.43.192:48089;branch=z9hG4bK4e1ab219cbde6190;rport
outgoing invite:
SIP/2.0/TCP 192.98.102.10;branch=z9hG4bK27b9.597e97c43da7ddd6be8520b662d616b9.0;i=5
SIP/2.0/TCP 192.168.43.192:48089;received=192.98.102.10;branch=z9hG4bK4e1ab219cbde6190;rport=49582
is it somehow possible to find out in the config script, what will the
branch value of the new via header be?
if that value would be known, it could be assigned to rtpengine
extra_id_pv pseudo variable before making the offer and could then be
used in rtpengine_delete to delete the right branch.
if the new via branch value cannot be known, any suggestions on how to
uniquely identify the outgoing branched of the invite?
-- juha
I am trying to setup a really simple (I hope so) IMS platform, with
some basic services (like voicemail, IVR, videoconference), but I'm kind
of stuck.
I have installed Kamailio IMS on debian and a OpenIMS VM also. But my
little problem begins when I try to configure a application server (in
this case, a TAS). I have tried with Elastix 3.0 (kamailio+asterisk),
asterisk alone, and read some about Restcomm.
For Elastix and asterisk, I get stuck trying to setup the trigger
point. For Restcomm, I only find documentation for integration with
Clearwater IMS.
So, I need your help. Somebody has something like that working? What
TAS (or components for a TAS) do you recommend? Tutorials, examples,
manuals that you know can help me?
Thanks for reading, and many many thanks for any advice.
David
Hi all,
I am answering a SIP call made to my softphone. The softphone rings and also sends a 200 OK correctly. But the other end that made the call is seeing a failure in the SDP of the 200 OK packet.
This error is seen in SDP header => Invalid SDP line (no '=' delimiter) error
w.r.t w_frame4_public_ip.pcap It looks like the failure as seen in wireshark is because line "audio 5000 RTP/AVP 8 101" should really be "m= audio 5000 RTP/AVP 8 101"
so it doesn't see the '=' delimiter.
But if you look at w_frame52_local_ip.pcap, the packets sent from the local IP has the contents correct.
It is only after passing the public IP and at the network premises, that this failure is reported by the Session Border controller.
I am not able to understand why a correctly formed packet sent out is seen "corrupted" at the network end.
Any ideas really appreciated.
Thanks,
Badri.
Hello,
I am trying to forward the register to multiple Asterisk using Kamailio.
The basic Idea is:
Softphone A---->Kamailio-----> Asterisk 1
------> Asterisk 2.
I follow Asipto howto http://lylix.net/kamailio, but the problem I see in
my case, is that registration IP in Asterisk is Kamailio´s IP. This causes
calls going from kamailio to asterisk and then back to kamailio. In my
scenario, I need that softphone A IP appeard in Asterisk realtime contact
IP.
The call path would be:
Softphone B -> kamailio -> Asterisk 1 or Asterisk 2 (depending on
dispatcher) -> Softphone A
Following Asipto how to the call goes Softphone A -> kamailio -> asterisk
-> kamailio Softphone B
Hope I explain myself clearly.
Thank you in advance.