Hello,
I am trying to set up a WebRTC2SIP Gateway by using Kamailio and rtpengine.
So far, everything is working fine, I'm able to register an extension and
make a call, but for some reason, when i'm trying to call a WebRTC
extension from any SIP Extension Kamailio is sending INVITE, WebRTC
extension is sending back 200 OK, and then Kamailio is trying to send an
ACK through UDP protocol, and not through wss, as it's supposed to do. This
is how invite is looking:
INVITE sip:nl7oe4ss@vjbh7r4im6j7.invalid;transport=wss SIP/2.0
Record-Route: <sip:my-company.net
;transport=udp;ftag=as1789445c;lr=on;nat=yes>
Via: SIP/2.0/WSS 123.123.123.123:10443
;branch=z9hG4bKe655.29d7c135a302f3eb803902d4f5a8da7e.0
Via: SIP/2.0/UDP 192.168.50.237:5060
;received=192.168.50.237;branch=z9hG4bK7d2e534e;rport=5060
Max-Forwards: 70
From: "WebRTC" <sip:11@my-company.net>;tag=as1789445c
To: <sip:15@192.168.50.210:5060>
Contact: <sip:11@192.168.50.237:5060>
Call-ID: 7fc800de060197fa2315c93763873092(a)my-company.net
CSeq: 102 INVITE
User-Agent: Proxy
Date: Wed, 03 Apr 2019 17:11:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Alert-Info:
Content-Type: application/sdp
Content-Length: 596
Server: SIP Proxy
and then WebRTC app is replying with 200 OK:
SIP/2.0 200 OK
Record-Route: <sip:my-company.net
;transport=udp;ftag=as1789445c;lr=on;nat=yes>
Via: SIP/2.0/WSS 123.123.123.123:10443
;branch=z9hG4bKe655.29d7c135a302f3eb803902d4f5a8da7e.0
Via: SIP/2.0/UDP 192.168.50.237:5060
;received=192.168.50.237;branch=z9hG4bK7d2e534e;rport=5060
To: <sip:15@192.168.50.210:5060>;tag=dk4fa8ftt6
From: "WebRTC" <sip:11@my-company.net>;tag=as1789445c
Call-ID: 7fc800de060197fa2315c93763873092(a)my-company.net
CSeq: 102 INVITE
Contact: <sip:nl7oe4ss@vjbh7r4im6j7.invalid;transport=wss>
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: outbound
User-Agent: Proxy-WEBRTC
Content-Type: application/sdp
Content-Length: 901
and finally, Kamailio is trying to send this ack through UDP protocol:
ACK sip:nl7oe4ss@22.22.22.22:57421;transport=wss SIP/2.0
Via: SIP/2.0/UDP 192.168.50.237:5060;branch=z9hG4bK56363ddf;rport
Route: <sip:my-company.net;transport=udp;ftag=as1789445c;lr=on;nat=yes>
Max-Forwards: 70
From: "WebRTC" <sip:11@my-company.net>;tag=as1789445c
To: <sip:15@192.168.50.210:5060>;tag=dk4fa8ftt6
Contact: <sip:11@192.168.50.237:5060>
Call-ID: 7fc800de060197fa2315c93763873092(a)my-company.net
CSeq: 102 ACK
User-Agent: Proxy
Content-Length: 0
If i'm trying to force it through TLS, i'm receiving error:
get_send_socket2(): protocol/port mismatch (forced tls:123.123.123.123:10443,
to udp:22.22.22.22:23317)
Can someone point me in the right direction, please?
Thank you.
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 All,
Below is the syntax am using to print the line number in kamailio
configuration file.
xlog("L_WARN","Hi Mahesh at line $cfg(line)\n");
Unfortunately , when i do this looks like kamailio is not starting, Any
help please regarding what am i missing here. I tried Xlogl as well, which
doesnt print the line number by default in my setup.
This is the details i had read to understand config file attributes :
$cfg(key) - Config File Attributes
Attributes related to configuration file.
The key can be:
- line - return the current line in config
- name - return the name of current config file
- file - return the name of current config file
- route - return the name of routing block
Example:
send_reply("404", "Not found at line $cfg(line)")
Hello!
I have notice some issues with lookup(aliases) since I upgraded to 5.2 from
5.1:
First with kamctl: If this default command is executed:
ctl_cmd_run ul.add "$USRLOC_TABLE" "$OSERUSER@$OSERDOMAIN" "$2" \
"$UL_EXPIRES" "$DEFAULT_Q" "$UL_PATH" "$UL_FLAGS" "$BR_FLAGS" "$ALL_METHODS"
as $ALL_METHODS is not defined, there is a huge value in methods and then
lookup(aliases) fails with -2.
I have to update kamctl script and put -1 ("-1") instead of the variable.
So, now, methods is a NULL value in DB and my alias is not ignored. But I
use to make my alias like this: sip:number@blabla_1.local
Which is no longer works: ERROR: registrar [common.c:62]: extract_aor():
failed to parse AoR [sip:number@blabla_1.local]
Any idea of the difference since 5.2 with this behaviour? Without the
underscore, the AoR seems to be valid but I need to support underscore as
before.
Regards,
Igor.
Hello,
I've been looking at the sip_p_charging_vector() function of siputils. From
the documentation it is not evident how the PCV header is generated. Are
the exported $pcv pseudovariables writable, so that the function canl use
those when generating the header? Thanks!
BR,
George
Hi ,
I have a query regarding Kamailio open source SIP server.
Does it have any release, where its supports the feature of IMS supporting
VOLTE with real commercial mobile phones(IMSI based) ?
If so, what is the release number ? Please also let me know about the
supported mobile phones list.
BR,
Shubhendu
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Hello list,
Hope you all doing well!
I am using the ACC module and using the setflag() function as done in
several examples. It works fine. However, I've added the t_newtran()
function almost in the begging of the INVITE handler to help the
retransmission detection and after that I noticed the ACC was not saving
anything in DB.
So after debugging I discovered that if I call the t_newtran() before
setting the ACC flags, the module will not save the calls in DB, but if I
call it after setting the ACC flags, it works....
So my question is, is this a bug or it is a expected side effect so when
one is using t_newtran you must be careful and set all your transaction
flags before? (ACC are the only transaction flags I am using so can't tell
if other modules have the same problem)
This is happening in Kamailio 5.2.2.
Thank you!
Kind regards,
Patrick Wakano
Hello,
I have created a "How-To" blog post on using Kamailio as "session border controller" for Microsoft Teams Direct Routing:
https://skalatan.de/de/blog/kamailio-sbc-teams
You can this way use all the existing possibilities that Kamailio provide to interact with MS Teams as well.
Best regards,
Henning
--
Henning Westerholt - https://skalatan.de/blog/
Kamailio services - https://skalatan.de/services
Maxim Malygin <maxim.malygin(a)gmail.com>
17:08 (5 минут назад)
кому: sr-users
Hello,
I need to form list of Contact headers with "q" parameters for "302 Moved
Temporary".
I do something like this:
...
# $xavp(contacts) - contains list of contact URIs (uri) and Q values (q)
$var(i) = 0;
$var(num) = $cnt($xavp(contacts));
while($var(i) < $var(num)) {
if($var(i) == 0) {
# How to add q value to this branch?
$ru = $xavp(contacts[$var(i)]=>uri);
} else {
append_branch("$xavp(contacts[$var(i)]=>uri)",
"$xavp(contacts[$var(i)]=>q)");
}
}
send_reply("302", "Moved Temporarily");
...
Is it possible to set "q" value for the main branch?
For additional branches I do it via append_branch("uri", q_value) but I
didn't find any way to do it for the main branch pointed via $ru/$du. The
pseudo-variable "$branch" also doesn't allow me to access the main branch.
Thanks,
Maxim