Hello the list,
I have a problem on the proxy with the audio between two calls bridged by a UAC.
When I made a normal call, no problems.
My UAC is nated.
UAC > Router > KAMAILIO
Frames arrives with private IP in the SDP.
U 2018/01/18 21:50:16.798581 217.112.180.235:1024 -> 217.112.180.10:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 217.112.180.10;branch=z9hG4bK2d06.2f2a1856bde881173a3fc413c4136b83.0.
Via: SIP/2.0/UDP 84.14.241.179:5060;rport=5060;branch=z9hG4bK0dBe3143bcf7f60a70b.
Record-Route: <sip:217.112.180.10;lr=on;ftag=gK0d4dfe6f;did=227.c0d2;nat=yes>.
From: <sip:32XXXXXX87@ >;tag=gK0d4dfe6f.
Call-ID: 940401290_111374574(a)84.14.241.179.
CSeq: 29328 INVITE.
Contact: <sip:32XXXXXX61@192.168.2.2:5060;transport=udp>.
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY.
Supported: timer,100rel.
Server: IP Office 10.1.0.0.0 build 237.
Min-SE: 1800.
Require: timer.
Session-Expires: 1800;refresher=uas.
To: <sip:32XXXXXX61@217.112.180.235>;tag=998e429e819ba686.
Content-Type: application/sdp.
Content-Length: 202.
.
v=0.
o=UserA 3721571424 1025404311 IN IP4 192.168.2.2.
s=Session SDP.
c=IN IP4 192.168.2.2.
t=0 0.
m=audio 49154 RTP/AVP 9 101.
a=rtpmap:9 G722/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
I don't understand why but the proxy (in case of an incoming call) succeed to determine the public IP.
Jan 18 21:40:08 proxy1 rtpproxy[1314]: INFO:rxmit_packets:1fb3f1c6c8dfe738221842406be48450: caller's address filled in: 217.112.180.235:49154 (RTP)
Jan 18 21:40:08 proxy1 rtpproxy[1314]: INFO:rxmit_packets:1fb3f1c6c8dfe738221842406be48450: guessing RTCP port for caller to be 49155
Jan 18 21:40:16 proxy1 rtpproxy[1314]: INFO:rxmit_packets:1fb3f1c6c8dfe738221842406be48450: caller's address latched in: 217.112.180.235:49155 (RTCP)
Don't know why because the only information in SDP is 192.168.2.2
The Kamailio didn't send the information of the proxy to the UAC , but to the other end as this is an incoming call.
So I have audio in this case.
When I setup a bridge on the UAC to a second number, we have an issue. ( Kamailio 4.4.6 )
This is the same frames
U 2018/01/18 21:51:26.607270 217.112.180.235:1024 -> 217.112.180.10:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 217.112.180.10;branch=z9hG4bK4181.bb7aab84b6fb0aea5836b4d8874406ec.0.
Via: SIP/2.0/UDP 84.14.241.179:5060;rport=5060;branch=z9hG4bK0bB16ee5a3d300fee16.
Record-Route: <sip:217.112.180.10;lr=on;ftag=gK0b5d7652;did=cc6.5452;nat=yes>.
From: <sip:32XXXXXX87@ >;tag=gK0b5d7652.
Call-ID: 940248136_13287504(a)84.14.241.179.
CSeq: 21946 INVITE.
Contact: <sip:32XXXXXX61@192.168.2.2:5060;transport=udp>.
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY.
Supported: timer,100rel.
Server: IP Office 10.1.0.0.0 build 237.
Min-SE: 1800.
Require: timer.
Session-Expires: 1800;refresher=uas.
To: <sip:32XXXXXX61@217.112.180.235>;tag=559c99f5edcab5d4.
Content-Type: application/sdp.
Content-Length: 202.
.
v=0.
o=UserA 1781830446 4071482272 IN IP4 192.168.2.2.
s=Session SDP.
c=IN IP4 192.168.2.2.
t=0 0.
m=audio 49156 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
But in this case no audio
RTCP detected but no the RTP.
He took the private ip address "192.168.2.2" and this is the reason of the "no audio".
Jan 18 21:51:26 proxy1 rtpproxy[1314]: INFO:rtpp_command_ul_handle:940248136_13287504@84.14.241.179: lookup on ports 11264/10446, session timer restarted
Jan 18 21:51:26 proxy1 rtpproxy[1314]: INFO:rtpp_command_ul_handle:940248136_13287504@84.14.241.179: pre-filling callee's address with 192.168.2.2:49156
Jan 18 21:51:26 proxy1 rtpproxy[1314]: INFO:rxmit_packets:e1e387824e0753522b7bc55e02090784: callee's address latched in: 79.137.49.139:39176 (RTP)
Jan 18 21:51:32 proxy1 rtpproxy[1314]: INFO:rxmit_packets:e1e387824e0753522b7bc55e02090784: caller's address filled in: 217.112.180.235:49155 (RTCP)
Jan 18 21:51:32 proxy1 rtpproxy[1314]: INFO:rxmit_packets:940248136_13287504@84.14.241.179: callee's address filled in: 217.112.180.235:49157 (RTCP)
I would like to understand why with the first call, no issues to determine the RTP IP and not in the second case,
Hi all
Could anyone tell me where the current SEMS site is? iptel seem to have lost all references and moved their site to wix. Everything search wise points me back at iptel :-(
Or is the copy on github it at the moment? https://github.com/sems-server <https://github.com/sems-server>
Cheers
Mark
Hi list,
I'm using dispatcher module and I'm not able to figure it out how to
accomplish a simple task regarding gateways availability and probings.
My goal is to probe with SIP OPTIONS my three gatways every, for
example, 30 seconds. When one, or more, of them respond with unexpected
message or simply do not respond at all, dispatcher module put it/them
as inactive automatically.
I tried several conf scenarios but without success, all with gateway A
off. Fore example:
1 -> I configure the three gateways (A, B and C) in conf file
(sip.peers as configuration bellow) with flags 8 setted (probing
destination).
- When I start kamailio with probing mode equal to 1, all
gateways present flags AP (using kamctl dispatcher dump) and nothing
change (I only see OPTIONS sent to host B and C).
- with probing mode setted to 0, all gateways start with flags
AP, after 30 seconds, first OPTIONS are sent (only for B a C again),
both B and C pass to AX and after more 30 seconds (ping interval
defined) gateway A pass also to AX and OPTIONS stop being sent.
2-> Now I configured the gateways in conf file with flags 1 (inactive
destination).
- With prob mode 1, all started as IX, 30 seconds later it
sends OPTIONS only for gw B and C and both change state to AP and A
maintain IX. After more 30 seconds, no more OPTIONS are sent and gw A
pass also to AP (hoped it remained IP or IX but not happened).
I also tried other combinations of these two parameters, but only the
onde I described above make me sense for my purpose.
Maybe the solution is simple but I'm not getting there or maybe this it
is only possible using failure or timeout routes and setting flags
manually to gateways :(
Can anyone point out the solution for this if it exists?
module configuration used:
modparam("dispatcher", "list_file", "sip.peers");
modparam("dispatcher", "flags", 2);
modparam("dispatcher", "dst_avp", "$avp(AVP_DST)");
modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)");
modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)");
modparam("dispatcher", "sock_avp", "$avp(AVP_SOCK)");
modparam("dispatcher", "ds_ping_method", "OPTIONS");
modparam("dispatcher", "ds_ping_from", "sip:ping@test.tt");
modparam("dispatcher", "ds_ping_interval", 30);
modparam("dispatcher", "ds_probing_threshold", 1);
modparam("dispatcher", "ds_inactive_threshold", 1);
modparam("dispatcher", "ds_ping_reply_codes",
"class=2;class=4;class=5;class=3");
modparam("dispatcher", "ds_probing_mode", 1);
Thanks in advance,
Paulo
--
-------------------------------------------
Paulo Ferreira
VoIP@RCTS - Área de Infraestruturas Aplicacionaisz
FCCN
http://www.fccn.pt/
Av. do Brasil, n.º 101
1700-066 Lisboa - Portugal
Telefone|Phone: +351 218440100; Fax: +351 218472167
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Hey!
I'm trying to use the KeepAlive options avaliable in the Dialog modules.
For that I use the function dlg_set_property() like this :
dlg_set_property("timeout-noreset");
dlg_set_property("ka-dst");
dlg_set_property("ka-src");
However, kamailio can't seem to get the answers from "ka-src". I've tried
several scenarios and different call directions and i can never get the
answer from the "ka-src". Is there any known issue about this feature ?
What arguments kamailio uses to build the OPTIONS request? I think i might
be albe to solve the issue if i know this.
I have some endpoints behind NAT, Kamailio has multiple IP's and i do
number normalization so my proxy works in E.164. I've made some tests and
those features don't seem to be the problem, but i cannot be sure. The
"ka-dst" works all the time under all situations.
Thanks in advance,
Cheers
Hello,
I am testing failover scenarios between two Active/Active kamailios for Sip
over WebSockets and WebRTC.
In my test scenario, I have setup the call through proxy-a, then stopped
the kamailio service so that the sip client reconnects to proxy-b. Dialogs
and registrations are shared across both instances using MariaDB and
Galera. When the SIP client reconnects to proxy-b, I send a BYE and see the
following errors in the logs:
Jan 18 09:12:36 webrtc-proxy-b /usr/sbin/kamailio[6975]: ERROR: <script>:
Connection ID: 1
Jan 18 09:12:36 webrtc-proxy-b /usr/sbin/kamailio[6975]: ERROR: <script>:
WITHINDLG
Jan 18 09:12:36 webrtc-proxy-b /usr/sbin/kamailio[6975]: ERROR: <script>:
RELAY
Jan 18 09:12:36 webrtc-proxy-b /usr/sbin/kamailio[6975]: WARNING: <core>
[core/msg_translator.c:2765]: via_builder(): TCP/TLS connection (id: 0) for
WebSocket could not be found
Jan 18 09:12:36 webrtc-proxy-b /usr/sbin/kamailio[6975]: ERROR: <core>
[core/msg_translator.c:1980]: build_req_buf_from_sip_req(): could not
create Via header
Jan 18 09:12:36 webrtc-proxy-b /usr/sbin/kamailio[6975]: ERROR: tm
[t_fwd.c:476]: prepare_new_uac(): could not build request
Jan 18 09:12:36 webrtc-proxy-b /usr/sbin/kamailio[6975]: ERROR: tm
[t_fwd.c:1732]: t_forward_nonack(): failure to add branches
Jan 18 09:12:36 webrtc-proxy-b /usr/sbin/kamailio[6975]: ERROR: sl
[sl_funcs.c:362]: sl_reply_error(): stateless error reply used: No error
(2/SL)
It seems that core/msg_translator.c is using the wrong WebSocket connection
ID (0 rather than 1). Is this a bug in Kamailio or a limitation that I do
not understand? I have assumed that the BYE is a new transaction for the
existing dialog so t_relay() should be able to forward the message. I can
provide config and any additional error logs if required.
Any help is much appreciated.
Thanks,
James
exec_msg gives the following warning. Even when no variables are used in the parameters
8(29619) WARNING: exec [exec_hf.c:464]: append_fixed_vars(): uri not parsed
8(29619) WARNING: exec [exec_hf.c:482]: append_fixed_vars(): orig URI not parsed
Hi K-team,
I am processing incoming INVITE events in my lua script.
To print the logs with kamailio syslog, i am using
sr.log("info", "-- Collecting data from kamailio -- ")
I am searching for a way to send selective logs for an incoming call from
lua script through HEP3 towards Homer server.
How can I make it work? What utility or library is helpful.
*PS:* captagent is configured on the same server for SIP traces.
--
regards,
abdul basit