Hello,
I'm using RTPproxy for the first time in bridged mode and I can't get kamailio/rtpproxy to rewrite the c parameter to the correct public ip address of kamailio.
The setup is as following:
Carrier ------[fiber]------ Kamailio ---------[lan]--------- Freeswitch
Kamailio is listening on two interfaces:
1) Private: 172.0.0.1
2) Public: 192.168.0.1 (since we have a dedicated fiber with our carrier, this is its public address)
Freeswitch is listening on:
1) 172.0.0.2
Carrier is on:
1) 10.0.0.1
I've started an rtpproxy instance on the Kamailio box using:
rtpproxy -s udp:127.0.0.1:7721 -u rtpproxy rtpproxy -p /var/run/rtpproxy/rtpproxy.pid -l 192.168.0.1 172.0.0.1
I've played around with rtpproxy_manage() and the various flags (ie, ei), but I can't get kamailio to set the correct public IP when the 200 OK has to be sent back to the carrier.
It always sets it to its private address, instead of its public address.
I'm using Kamailio 4.2 with sippy/rtpproxy 2.0.
Could someone please point me into the right direction?
Thanks!
Grant
Dear friends,
I am working on a program on Kamailio and rtpengine proxy. I am wondering whether can I set Kamailio and rtpengine daemon on different physical machines. For example, I set Kamailio on a machine with IP address:10.109.247.80, and launch rtpengine daemon on another machine with interface parameter as 10.109.247.90 and ng port 7723. I set parameter in Kamailio.cfg with modparam(“rtpengine”, “rtpengine_sock”, “udp:10.109.247.90:7723”).
Unfortunately I got debug message like this:
ERROR: rtpengine [rtpengine.c:1710]: send_rtpp_command(): can't send command to a RTP proxy
ERROR: rtpengine [rtpengine.c:1746]: send_rtpp_command(): proxy <udp:10.109.247.90:7723> does not respond, disable it
ERROR: rtpengine [rtpengine.c:1616]: rtpp_test(): proxy did not respond to ping
And, I also tried to set Kamailio and rtpengine daemon in a same machine,and use modparam(“rtpengine”, “rtpengine_sock”, “udp:localhost:7723”). And Kamailio can work functionally under this situation. rtpengine daemon can receive ping message from Kamailio and rtpengine daemon can work as suspected. So for the later case, is it supposed that Kamailio be in the same machine with same localhost address? Otherwise, what’s the reason for my ERROR?
------------------------------------
北京邮电大学网络技术研究院
网络与交换技术国家重点实验室
田军
+86 18810315790
mozillafire(a)bupt.edu.cn
------------------------------------
Hi,
I'm trying to set up a Kamailio (4.1.3) server which converts between IPv6
and IPv4. Everything looks pretty good. Now I have a test client which
sends packets via IPv6, but in contact header, SDP etc. there are still
IPv4 addresses. (This comes from some converting from v4 to v6 earlier.)
Thus, Kamailio tries to do a fix_nated_contact(). But after fixing, the
Contact URI is wrong.
Example:
inbound
Contact: <sip:bob@192.168.8.132>
outbound
Contact: <sip:bob@1234:1234:0:1234:0:0:0:2>
The correct way of transforming the Contact URI would be:
Contact: <sip:bob@[1234:1234:0:1234:0:0:0:2]>
This incorrect Contact URI doesn't hurt first, but when the called party
wants to hang up the call, it addresses the URI from Contact header, and
Kamailio can't parse the Request URI:
Jan 22 12:03:48 router /usr/sbin/kamailio[21309]: ERROR: pv
[pv_core.c:304]: pv_get_ruri_attr(): failed to parse the R-URI
Jan 22 12:03:48 router /usr/sbin/kamailio[21309]: ERROR: rr [loose.c:934]:
loose_route(): failed to parse Request URI
Jan 22 12:03:48 router /usr/sbin/kamailio[21309]: ERROR: domain
[domain.c:140]: is_uri_host_local(): error while parsing R-URI
Is this a bug in the nathelper module? Was it never meant to handle IPv6
addresses? Or do I understand something wrong?
Best Regards,
Sebastian
Hi All,
I have a strange issue, I have set the module parameter for nathelper's
force_socket to a specfic ip/port, however, when I perform a sip trace I
can see that all locally generated options messages are not sent from
the socket defined in the modules parameters.
I am not setting $fs anywhere in the script, so I am not over-writing
it. Do any of the other module parameters have a bearing on if nathelper
uses the force_socket parameter?
My current nathelper settings are:
modparam("nathelper", "received_avp", "$avp(RECEIVED)")
modparam("nathelper", "natping_interval", 20)
modparam("nathelper", "natping_processes", 4)
modparam("nathelper", "ping_nated_only", 1)
modparam("nathelper", "sipping_from", "sip:keepalive@domain.com")
modparam("nathelper", "sipping_method", "OPTIONS")
modparam("nathelper", "sipping_bflag", NAT_BFLAG)
modparam("nathelper", "force_socket", "1.2.3.4:5060")
I am using kamailio v4.3.1:
# kamailio -V
version: kamailio 4.3.1 (x86_64/linux) f38e67
flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS,
DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC,
F_MALLOC, DBG_F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT,
USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST,
HAVE_RESOLV_RES
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
id: f38e67
compiled on 18:15:23 Jul 20 2015 with gcc 4.4.7
Thanks
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
Hello,
We're testing kamailio 4.3.4 with two kamailio's communicating with each other. One of them is a gateway with rtpengine at the carrier side, the other is a basic sip router.
When calling a destination and cancelling the INVITE a 487 is received from the carrier and directly ACK'd by the first kamailio and in turn is forwarded to the second.
But the 487 response received by the 2nd kamailio doesn't have a Via header which results in kamailio printing the error: receive_msg(): no via found in reply.
Any ideas why the forwarded 487 doesn't contain a Via header and how to fix this?
Note: IP's are changed.
Carrier: 10.0.0.1
Kamailio 1: 10.14.0.1
Kamailio 2: 10.14.0.2
U 10.0.0.1:5060 -> 10.14.0.1:5060
SIP/2.0 487 Request Cancelled.
Via: SIP/2.0/UDP 10.14.0.1;branch=z9hG4bK8e28.642c68037d059634308d4be8fe3f0cdc.0.
From: "+31612345678" <sip:+31612345678@10.14.0.3:5060>;tag=109ac722.
To: <sip:003112345678@10.14.0.3:5060>;tag=sbcsipuas_1_C22196_20160314080307791_b59sb10.
Call-ID: 78100MjY3OGExZmMyODU2MzAyMzM5ODA3MmNhMDdlMzg0MzU.
Contact: <sip:10.0.0.1:5060>.
CSeq: 2 INVITE.
Server: sbc_5.
Content-Length: 0.
.
#
U 10.14.0.1:5060 -> 10.0.0.1:5060
ACK sip:003112345678@10.0.0.1:5060 SIP/2.0.
Via: SIP/2.0/UDP 10.14.0.1;branch=z9hG4bK8e28.642c68037d059634308d4be8fe3f0cdc.0.
Max-Forwards: 67.
To: <sip:003112345678@10.14.0.3:5060>;tag=sbcsipuas_1_C22196_20160314080307791_b59sb10.
From: "+31612345678"<sip:+31612345678@10.14.0.3:5060>;tag=109ac722.
Call-ID: 78100MjY3OGExZmMyODU2MzAyMzM5ODA3MmNhMDdlMzg0MzU.
CSeq: 2 ACK.
Content-Length: 0.
.
#
U 10.14.0.1:5060 -> 10.14.0.2:5060
SIP/2.0 487 Request Cancelled.
From: "+31612345678" <sip:+31612345678@10.14.0.3:5060>;tag=109ac722.
To: <sip:003112345678@10.14.0.3:5060>;tag=sbcsipuas_1_C22196_20160314080307791_b59sb10.
Call-ID: 78100MjY3OGExZmMyODU2MzAyMzM5ODA3MmNhMDdlMzg0MzU.
Contact: <sip:10.0.0.1:5060>.
CSeq: 2 INVITE.
Server: sbc_5.
Content-Length: 0.
Hi,
I have two SIP servers, *A* and *B*, connected each other though a OPENVPN
tunnel. The server *B* needs to t_relay() every SIP message containing the
method MESSAGE to the server *A* but these messages never reach destination.
I have tested the tunnel connectivity and works fine. I wrote a Perl script
(located in *B*) that sends SIP MESSAGES to Kamailio (located in *A*) trying
to figure out what is happening but these messages are received by *A* and
processed correctly but when *B* does the same from Kamailio, it is never
received.
Here is the route part of kamailio.cfg in *B*:
Observation: ($rU == "1004") result is *true*
*
if(is_method("MESSAGE"))
{
if($rU == "1004")
{
xlog("L_INFO","En 1004");
rewritehost("10.8.0.1");
if (!t_relay())
xlog("L_INFO","MIO Error en t_relay");
t_reply("200", "Ok");
xlog("L_INFO","MIO despues rewrite");
exit;
}
.....*
The perl script that WORKS:
*$msg = 'MESSAGE sip:1004@192.168.1.2:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2
From: "2002" <sip:2002@192.168.1.2 <sip%3A2002(a)192.168.1.2>>;tag=1837944796
To: <sip:1004@192.168.1.2 <sip%3A1004(a)192.168.1.2>>
Call-ID: 19722852989(a)192.168.1.2
CSeq: 15773 MESSAGE
Contact: <sip:2002@**PublicIP-protected**:5060>
Max-Forwards: 29
User-Agent: DBL
Content-Type: text/plain
Content-Length: 34
+595981[protected]
hello from kamailio
';
use IO::Socket;
my $sock = IO::Socket::INET->new( Proto=>'udp',
PeerHost=>'192.168.2.102',
PeerPort=>'5060');
print "Sending msg $msg\n";
$sock->send($msg) or die "error sending $!\n";
*Please help!
Thanks in advance.
Carlos.
Hi,
About 3 weeks ago i upgraded one of my production server with latest stable
kamailio version 4.2.1-fad00a. Now i am getting a lot of complaints about
missing CDR events in ACC table. I observe following problems,
1. There are only BYE records in acc table, no record for INVITE or ACK.
2. In kamailio logs when ACK is received against 200 OK response for
INVITE, i see following errors,
--
ERROR: <core> [parser/parse_from.c:113]: parse_from_uri(): failed to parse
>From uri
ERROR: pv [pv_core.c:434]: pv_get_xto_attr(): cannot parse From URI
NOTICE: <script>: [udp:<null>@1.0.0.127:5060]: Call from 'you(a)kamailio.org'
to 'you(a)kamailio.org' has been hanged up by '<null>' at '1419364717.255484'
--
Of course all these errors are bogus, I have checked all headers in ACK
(not just FROM header), they all seem perfectly fine and valid.
3. Then the dialog times out,
--
WARNING: dialog [dlg_handlers.c:1440]: dlg_ontimeout(): timeout for dlg
with CallID '6D8BD23CAC65AE3C1DE1D0B531F87B8CFEAA9CB9' and tags
'1D3ECD34F5731AB845BA3064AC95BB2D'
'7f55e81e0630-100007f-13c4-6009-2440a4-5fa31570-2440a4'
--
4. Any further sequential requests complain about "unable to find dialog",
e.g.
--
NOTICE: <script>: Sequencial 'BYE' request received from caller
ERROR: uac [replace.c:591]: restore_uri(): new URI [] shorter than old URI [
sip:00xxxxxxxxxx@sip.domain.com]
WARNING: dialog [dlg_handlers.c:1174]: dlg_onroute(): unable to find dialog
for BYE with route param '5ae1.d595' [7845:22877]
--
5. However the acc record for BYE is written to db and log file,
--
NOTICE: acc [acc.c:318]: acc_log_request(): ACC: transaction answered:
timestamp=1419364760;method=BYE;from_tag=7f55e81e0630-100007f-13c4-6009-2440a4-5fa31570-2440a4;to_tag=1D3ECD34F5731AB845BA3064AC95BB2D;call_id=6D8BD23CAC65AE3C1DE1D0B531F87B8CFEAA9CB9;code=200;reason=OK;src_user=00xxxxxxxxxx;src_domain=
sip.domain.com
;src_ip=xx.xx.xx.xx;dst_ouser=+1xxxxxxxxxx;dst_user=1xxxxxxxxxx;dst_domain=yy.yy.yy.yy
--
The same config was working fine with older version 4.2.0-97cab8. The
kamailio config i am using is pretty much standard,
--
#!define FLT_ACC 1
#!define FLT_ACCMISSED 2
#!define FLT_ACCFAILED 3
#!define FLT_DLG 4
...
modparam("acc", "early_media", 1)
modparam("acc", "report_ack", 1)
modparam("acc", "report_cancels", 1)
modparam("acc", "detect_direction", 1)
modparam("acc", "log_flag", FLT_ACC)
modparam("acc", "log_missed_flag", FLT_ACCMISSED)
modparam("acc", "failed_transaction_flag", FLT_ACCFAILED)
# log to db
modparam("acc", "db_flag", FLT_ACC)
modparam("acc", "db_missed_flag", FLT_ACCMISSED)
modparam("acc", "db_url", "DBURL")
...
request_route {
# per request initial checks
route(REQINIT);
# NAT detection
route(NATDETECT);
# handle requests within SIP dialogs
route(WITHINDLG);
# CANCEL processing
if (is_method("CANCEL")) {
if (t_check_trans()) {
t_relay();
};
exit;
};
#### only initial requests (no To tag) ####
t_check_trans();
....
# account only INVITEs
if (is_method("INVITE")) {
setflag(FLT_DLG); # create dialog
setflag(FLT_ACC); # do accounting
setflag(FLT_ACCFAILED); # ... even if the transaction fails
$avp(dlg_timeout) = 60;
dlg_manage();
....
}
--
Any ideas why its happening? Since it is 3 weeks old so may be problem has
already been spotted and fixed by someone else. Otherwise let me know how
can i provide more info to help fix this issue.
Thank you.
Hi
I am novice to kamailio , I have installed kamailio-4.2.0 and SIREMIS. I
have add and registered two IP phones with kamailio these phone able to
call each other.
How can i make outbound call with that I am using a asterisk server as PSTN
Gateway. i am not famalier with kamailio script.
Regards
Sharad Tyagi
SparkTG info Pvt Ltd
Hi, I have Kamailio as SIP server and RTP server. Client is PJSIP.
I read that STUN is for non-symmetric NAT, and RTP server is for symmetric
NAT.
Supposed A calls B.
If A, B both use symmetric NAT and STUN, they cannot hear each other
If A or B use non-symmetric NAT and NOT using STUN, they cannot hear each
other.
Why is that?
I read http://tools.ietf.org/id/draft-takeda-symmetric-nat-traversal-00.txt
for Prediction Failure, is that related to this problem ?
--
Khoa Pham
HCMC University of Science
Faculty of Information Technology
