when i make call from UDP/TLS/RTP/SAVP baresip to RTP/AVP sems,
rtpengine gets called on initial invite/200 ok like this and audio works
fine:
Nov 18 02:46:06 rautu /usr/bin/sip-proxy[926]: INFO: ===== rtpengine_offer(ICE=force replace-session-connection replace-origin via-branch=1 RTP/AVP trust-address)
Nov 18 02:46:06 rautu /usr/bin/sip-proxy[878]: INFO: ===== rtpengine_answer(ICE=force via-branch=2 trust-address)
however, when baresip makes re-invite, rtpengine gets called using the
same offer/answer flags, but sems clears the call. on syslog, i see
this:
Nov 18 02:46:59 rautu /usr/bin/sip-proxy[919]: INFO: Routing in-dialog INVITE <sip:127.0.0.1:5090;transport=udp> from <sip:jh@test.tutpro.com>
Nov 18 02:46:59 rautu /usr/bin/sip-proxy[879]: INFO: ===== rtpengine_answer(ICE=force via-branch=2 replace-session-connection replace-origin)
Nov 18 02:46:59 rautu rtpengine[29718]: [abd2bcf75f71af57 port 50444] SRTP output wanted, but no crypto suite was negotiated
Nov 18 02:46:59 rautu /usr/bin/sip-proxy[919]: INFO: Routing in-dialog ACK <sip:127.0.0.1:5090;transport=udp> from <sip:jh@test.tutpro.com>
Nov 18 02:46:59 rautu rtpengine[29718]: [abd2bcf75f71af57 port 50462] Discarded invalid SRTP packet: authentication failed
Nov 18 02:46:59 rautu sems[20439]: [#b7193b70] [receive, AmRtpAudio.cpp:212] ERROR: decode() returned -4
Nov 18 02:46:59 rautu /usr/bin/sip-proxy[878]: INFO: ===== rtpengine_delete()
Nov 18 02:46:59 rautu /usr/bin/sip-proxy[878]: INFO: Routing in-dialog BYE <sip:jh-0x9a95b00@192.98.102.30:5066;transport=tcp> from <sip:jh@as.test.tutpro.com> to <sip:192.98.102.30:46718;transport=TCP> based on gruu
Nov 18 02:46:59 rautu rtpengine[29718]: [abd2bcf75f71af57 port 50463] Error parsing RTCP header: invalid packet type
and on baresip console this:
dtls_srtp: ---> DTLS-SRTP complete (audio/RTCP) Profile=AES_CM_128_HMAC_SHA1_80
dtls_srtp: incoming DTLS connect from 192.98.102.30:50524
dtls_srtp: verified sha-1 fingerprint OK
dtls_srtp: ---> DTLS-SRTP complete (audio/RTP) Profile=AES_CM_128_HMAC_SHA1_80
srtp: recv: failed to decrypt RTCP-packet (Unknown error 217)
srtp: recv: failed to decrypt RTCP-packet (Unknown error 217)
srtp: recv: failed to decrypt RTP-packet (Unknown error 217)
srtp: recv: failed to decrypt RTP-packet (Unknown error 217)
srtp: recv: failed to decrypt RTP-packet (Unknown error 217)
srtp: recv: failed to decrypt RTP-packet (Unknown error 217)
sip:jh@as.test.tutpro.com: session closed: Connection reset by peer
could it be that at some point during the re-invite, sems gets srtp audio
and therefore clears the call? if so, is it a bug in rtpengine?
-- juha
Hi!
Kamailio 4.1.6 with default settings for usrloc and registrar modul and
db_mode=1 (write through).
Fritzbox sends a reREGISTER and Kamailio tries to insert the contact
into the database, but this fails as this contact for this AoR already
exists.
So I think there are 2 scenarios: Kamailio false INSERTs instead of
UPDATEs the contact, or the contact was deleted from usrloc, but not
from the database.
Frankly I see the in the mysql logfile that immediately after the failed
INSERT the record is deleted from DB.
Does anybody have a pointer what might be the problem here? AFAIS the
client always uses the same Contact and the same call-id. only Cseq
increases.
Thanks
Klaus
rtpengine README lists these rtp protocol and profile flags:
+ RTP/AVP, RTP/SAVP, RTP/AVPF, RTP/SAVPF
rtpengine source code, however, seems to know also these DTLS ones:
daemon/call.c: .name = "UDP/TLS/RTP/SAVP",
daemon/call.c: .name = "UDP/TLS/RTP/SAVPF",
can they be used in rtpengine_offer/answer calls even when they are
missing from README?
-- juha
Hey Paul,
Thanks for your feedback!
In this case the call will come in via UDP (so it has 2 record route
headers). I can still play tricks like reading the record-route headers
but, I was in some way hoping that is the responsibility of
loose_route() function to properly use the right protocol out of route
headers since we are a proxy.
Cheers,
DanB
On 14.11.2014 12:00, sr-users-request(a)lists.sip-router.org wrote:
> Message: 3
> Date: Thu, 13 Nov 2014 16:55:18 +0530
> From: Varghese Paul<varghesepaul87(a)gmail.com>
> To: "Kamailio (SER) - Users Mailing List"
> <sr-users(a)lists.sip-router.org>
> Subject: Re: [SR-Users] UDP-TCP bridging in-dialog SUBSCRIBE
> Message-ID:
> <CAM33Ch4furD1FM-ZNKfbQ0nXNQd88BmzW73gjjb=cVyuC0K=8w(a)mail.gmail.com>
> Content-Type: text/plain; charset="utf-8"
>
> Hi Dan,
>
> You can check the transport value in the RURI and relay the SIP message as
> tcp.
>
> if ($*rP* == "tcp"){
> t_relay_to_tcp()
> }
>
> Regards
>
> Varghese Paul
This is my first time posting to the mailing list. I'm currently running
Kamailio 4.1, per the instructions on the website I got the tar bal for
Siremis 4.1 s well. I have followed the install instructions at
http://kb.asipto.com/siremis:install41x:main and I get all the way down to
the bottom where I need to login to the web interface and when I try to use
the default I get a very length openbiz error:
[{"target":"ERROR","content":"
\n[2014-11-17 11:51:15 (GMT)] An exception occurred while executing this
script:
\nError message: #8192, Non-static method BizSystem::dbConnection() should
not be called statically, assuming $this from incompatible context<\/font>
\nScript name and line number of error:
\/var\/www\/siremis-4.1.0\/openbiz\/bin\/data\/BizDataObj_Abstract.php:423<\/font>
\n
*function:<\/b> errorHandler ( 8192, \"Non-static method
BizSystem::dbConnection() should not be called...\",
\"\/var\/www\/siremis-4.1.0\/openbiz\/bin\/data\/BizDataObj_Abstract.php\",
423, Array(0) ) @ \/var\/www\/siremis-4.1.0\/openbiz\/bin\/sysheader.inc
117\nfunction:<\/b> userErrorHandler ( 8192, \"Non-static method
BizSystem::dbConnection() should not be called...\",
\"\/var\/www\/siremis-4.1.0\/openbiz\/bin\/data\/BizDataObj_Abstract.php\",
423, Array(0) ) @
\/var\/www\/siremis-4.1.0\/openbiz\/bin\/data\/BizDataObj_Abstract.php
423\nfunction:<\/b> getDBConnection ( ) @
\/var\/www\/siremis-4.1.0\/openbiz\/bin\/data\/BizDataObj_Lite.php
406\nfunction:<\/b> _run_search ( Array(2) ) @
\/var\/www\/siremis-4.1.0\/openbiz\/bin\/data\/BizDataObj_Lite.php
345\nfunction:<\/b> fetchRecords ( \"[username]='admin' and status='1'\",
Array(0), 1 ) @
\/var\/www\/siremis-4.1.0\/siremis\/modules\/service\/authService.php
98\nfunction:<\/b> authDBUser ( \"admin\", \"admin\" ) @
\/var\/www\/siremis-4.1.0\/siremis\/modules\/service\/authService.php
66\nfunction:<\/b> authenticateUser ( \"admin\", \"admin\" ) @
\/var\/www\/siremis-4.1.0\/siremis\/modules\/user\/form\/LoginForm.php
92\nfunction:<\/b> Login ( ) @
\/var\/www\/siremis-4.1.0\/openbiz\/bin\/BizController.php
310\nfunction:<\/b> invoke ( ) @
\/var\/www\/siremis-4.1.0\/openbiz\/bin\/BizController.php
110\nfunction:<\/b> dispatchRequest ( ) @
\/var\/www\/siremis-4.1.0\/openbiz\/bin\/BizController.php
32\nfunction:<\/b> include_once (
\"\/var\/www\/siremis-4.1.0\/openbiz\/bin\/BizController.php\" ) @
\/var\/www\/siremis-4.1.0\/siremis\/bin\/controller.php
6<\/div>\n------------------------------Please ask system administrator for
help...<\/div>\n"}]*
My web server is configured to use apache and I've checked the ownership on
the files and folder. I did a chmod 775 starting at the siremis-4.1.0/
directory. I did a search and found that the openbiz folder needs to be
right under the siremis-4.1.0/ folder which it is. I have tried
reinstalling several times only to get the same result. I'm sure I'm
missing something simple I just need to know what it is. I'm currently
running CentOS 7, Kernel 3.10.0-123.9.3, php 5.6.2-3, mariadb 10.0.14-1, &
httpd 2.4.6-18.
Dan
--
Hi All,
Looking for some help with a TCP issue in Kamailio.
I am running load testing through the proxy with a topology of UAC > Proxy
> UAS.
With UDP it works fine (using SIPp scripts) - when I turn to TCP as the
transport i get the following error in the kamailio logs:
ERROR: <core> [tcp_main.c:4159]: handle_tcpconn_ev(): connect
192.168.0.19:5060 failed
The IP address stated is the UAC which is sending an INVITE to the Proxy,
which seems to send back a 100 TRYING to the UAC which is followed by
repeated INVITE messages.
It looks like the leg from the Proxy to the UAS is never established using
TCP - however if I send the SIP messages to the Proxy via TCP and set the
UAS to receieve in UDP Kamailio completes with TCP > Kamailio > UDP.
When the UAC is set to send on TCP and UAS set to receive on TCP netstat
shows the ports as listening.
Can anyone please help?
Hello,
Sometimes kamailio with pipelimit crashes with segmentation fault, when CPU
load average >2.
The problem is not reproduced in the LAB (we used stress –c ).
Please find attached core file.
Thank you,
Julia
We have opensip configured with UAC auth to register SIP provide. I have
configured two provider and both got registered but very interesting thing
happened.
Provider "A" sending me 407 challenge for authentication - Working
Provider "B" sending me 401 for authentication - its failed to auth
what is the solution here? does UAC module work with 401?
Hi,
I have a wired problem in my Kamailio. Over WSS, Kamailio is
dropping the websocket connection right after it is getting the BYE from
other side. So the client side is not getting any BYE. Over normal http
it is working fine. Any idea why it might happen or where to look at?
Thanks in advanced!
