Hi.
I have 2 endpoints
1001 and 1002
1001 registered from 1 device
1002 registered from 2 devices at the same time
When I called to 1002 kamailio makes 2 branches
rtpengine_manage command called from branch rout for handling every branch
directly (it can be different endoint types (ws/tls/udp) for each branch)
When i picking up at the 1002 on one device server sends CANCEL (answered
elswhere) to another devices of 1002
rtpengine_manage reacts at the CANCEL and deletes this brach after 30
seconds
My question is:
What is solution for use rtpengine_manage and call branches together?
Hi guys,
I'm trying to append a content type to a message that has only one content
type. I tried using just set_body_multipart and now I'm using set_body
multipart and append_body_part with no success too.
The result is a message with multipart but has "two" sdp and no gtd. It has
the old sdp with no ip changed for rtpproxy and the new sdp with ip changed
BUT with no content type. And the message has not the end boundary
(--unique-boundary-1.)
Kamailio log has an info and a warning
Dec 26 11:36:22 dwrfsd01 ./kamailio[8847]: INFO: <core>
[msg_translator.c:1693]: get_boundary(): Content-Type hdr has no params
<application/sdp>
Dec 26 11:36:22 dwrfsd01 ./kamailio[8847]: WARNING: <core>
[msg_translator.c:1959]: build_req_buf_from_sip_req(): check_boundaries
error
kamailio.cfg
......
route[RELAY]{
$avp(ch) = 1
....
if ($avp(ch)){
set_body_multipart();
msg_apply_changes();
$var(gtd) ="IAM,\r\nGCI,asdafsdfasd\r\n\r\n";
append_body_part("$var(gtd)", "application/gtd",
"signal;handling=optional");
}
....
}
Ngrep
U 2016/12/26 11:36:22.496634 172.16.213.21:5060 -> 172.16.213.38:5060
...
Content-Type: application/sdp.
Content-Length: 137.
.
v=0.
o=user1 53655765 2353687637 IN IP4 172.16.213.21.
s=-.
c=IN IP4 172.16.213.21.
t=0 0.
m=audio 6000 RTP/AVP 8.
a=rtpmap:8 PCMA/8000.
U 2016/12/26 11:36:22.688458 172.16.213.38:5060 -> 172.16.208.11:5060
....
Content-Type: multipart/mixed;boundary="unique-boundary-1".
Mime-Version: 1.0.
Remote-Party-ID: <sip:1152565064@172.16.213.38:5060
>;party=calling;privacy=off;screen=no.
.
--unique-boundary-1.
Content-Type: application/sdp.
.
v=0.
o=user1 53655765 2353687637 IN IP4 172.16.213.21.
s=-.
c=IN IP4 172.16.213.21.
t=0 0.
m=audio 6000 RTP/AVP 8.
a=rtpmap:8 PCMA/8000.
.
--unique-boundary-1.
v=0.
o=user1 53655765 2353687637 IN IP4 172.16.213.38.
s=-.
c=IN IP4 172.16.213.38.
t=0 0.
m=audio 58724 RTP/AVP 8.
a=rtpmap:8 PCMA/8000.
a=sendrecv.
a=rtcp:58725.
a=ice-ufrag:hhXDwrco.
a=ice-pwd:5temIXP8veOGuD8Dsllm5HXkWU.
Any help will be very very appreciated!
Thanks in advance.
Diego
While waiting for Santa, it's a good time to also express my thanks and
greetings to all the friends, developers and community members that made
2016 another amazing year for Kamailio project.
Enjoy the winter holidays together with your beloved ones! Merry Christmas!
Daniel
--
Daniel-Constantin Mierla
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com
Friends,
Trying to calm down and prepare for Christmas with the family.
It’s been another great year for Kamailio and I’m proud to be part of the Kamailio development community.
We’ve made great releases, had a great conference and overall done good stuff :-)
For all of you that celebrate Christmas - a Merry Christmas! To the rest of you: Happy Holidays!
A big Thank You to all developers and a special giga-Thank You to Daniel for all the time you spend
working with the code, helping out in discussions on the lists and in the bug tracker.
And to all of you in the community - Thank You for being part of this!
Greetings from a cold Sweden, currently without snow in my area just north of Stockholm. We celebrate
on Christmas Eve (tomorrow) so there’s a lot of food being prepared right now and the last minute
shopping is just about to start...
Keep your Kamailio running!
/Olle
PS. If you are not currently following the project on Twitter - it’s time!
Find us as @kamailioproject
Hi,
my apologies for the off-topic posting but I think this could
be interesting for the SIP Community.
Baresip version 0.5.0 is now released, you can find the
announcement email here:
http://lists.creytiv.com/pipermail/re-devel/2016-December/001154.html
Merry Xmas from Norway!
/alfred
Hi list,
We have two eth, one with a private ip and the other with a public ip. We
have MHOMED configured. The call comes from private network and then is
routed to a public network via rtjson routing. The call is routed but the
thing is that the record route is being set with the private one (It's the
same when calls come from public and is routed to a private network, record
route has the public and not the private).
With mhomed configured shouldn't Record Route be updated as VIA header?
".... When activated, sip-router will select a socket that can reach the
destination (to be able to connect to the remote address). (sip-router
opens a UDP socket to the destination, then it retrieves the local IP which
was assigned by the operating system to the new UDP socket. Then this
socket will be closed and the retrieved IP address will be used as IP
address in the Via/Record-Route headers)..."
This is part of ngrep
U 2016/12/22 11:24:15.019643 172.16.213.21:5060 -> 172.16.213.38:5060
INVITE sip:11111111111@172.16.213.38:5060 SIP/2.0.
Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-13024-1-0.
From: "Test" <sip:2222222222@172.16.213.21:5060;user=phone>;tag=1.
To: <sip:1111111111@172.16.213.38:5060>.
Call-ID: 1-13024(a)172.16.213.21.
CSeq: 1 INVITE.
Contact: "2222222222 <sip:2222222222@172.16.213.21:5060>.
....
U 2016/12/22 11:24:15.023923 172.16.213.38:5060 -> 172.16.213.21:5060
SIP/2.0 100 trying -- your call is important to us.
Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-13024-1-0.
....
U 2016/12/22 11:24:15.038695 XXX.XXX.XXX.01:5060 -> XXX.XXX.XXX.02:5060
INVITE sip:1111111111 <011%201111-1111>@XXX.XXX.XXX.02:5060 SIP/2.0.
*-----> Record-Route: <sip:172.16.213.38;lr;ftag=1;did=1111>. <----*
Via: SIP/2.0/UDP XXX.XXX.XXX.01;branch=z9hG4bKc09f
Via: SIP/2.0/UDP 127.0.0.8;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.NLOBF6
This is my cfg
mhomed=1
request_route {
# per request initial checks
route(REQINIT);
# NAT detection
#route(NATDETECT);
# CANCEL processing
if (is_method("CANCEL")) {
if (t_check_trans()) {
route(RELAY);
}
exit;
}
# handle requests within SIP dialogs
route(WITHINDLG);
### only initial requests (no To tag)
# handle retransmissions
if(t_precheck_trans()) {
t_check_trans();
exit;
}
t_check_trans();
# authentication
route(AUTH);
# record routing for dialog forming requests (in case they are
routed)
# - remove preloaded route headers
remove_hf("Route");
if (is_method("INVITE|SUBSCRIBE")) {
record_route();
}
....
Thanks in advance.
Diego
2016-12-22 12:08 GMT-03:00 Diego Nadares <dnadares(a)gmail.com>:
> Hi list,
>
> We have two eth, one with a private ip and the other with a public ip. We
> have MHOMED configured. The call comes from private network and then is
> routed to a public network via rtjson routing. The is routed but the thing
> is that the record route is being set with the private one (It's the same
> when calls come from public and is routed to a private network, record
> route has the public and not the private).
>
> With mhomed configured shouldn't Record Route be updated as VIA header
>
> ".... When activated, sip-router will select a socket that can reach the
> destination (to be able to connect to the remote address). (sip-router
> opens a UDP socket to the destination, then it retrieves the local IP which
> was assigned by the operating system to the new UDP socket. Then this
> socket will be closed and the retrieved IP address will be used as IP
> address in the Via/Record-Route headers)..."
>
> This is part of ngrep
>
> U 2016/12/22 11:24:15.019643 172.16.213.21:5060 -> 172.16.213.38:5060
> INVITE sip:11111111111@172.16.213.38:5060 SIP/2.0.
> Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-13024-1-0.
> From: "Test" <sip:2222222222@172.16.213.21:5060;user=phone>;tag=1.
> To: <sip:1111111111@172.16.213.38:5060>.
> Call-ID: 1-13024(a)172.16.213.21.
> CSeq: 1 INVITE.
> Contact: "2222222222 <sip:2222222222@172.16.213.21:5060>.
> ....
>
> U 2016/12/22 11:24:15.023923 172.16.213.38:5060 -> 172.16.213.21:5060
> SIP/2.0 100 trying -- your call is important to us.
> Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-13024-1-0.
> ....
>
> U 2016/12/22 11:24:15.038695 XXX.XXX.XXX.01:5060 -> XXX.XXX.XXX.02:5060
> INVITE sip:1111111111 <011%201111-1111>@XXX.XXX.XXX.02:5060 SIP/2.0.
> *-----> Record-Route: <sip:172.16.213.38;lr;ftag=1;did=1111>. <----*
> Via: SIP/2.0/UDP XXX.XXX.XXX.01;branch=z9hG4bKc09f
> Via: SIP/2.0/UDP 127.0.0.8;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.NLOBF6
>
> This is my cfg
>
> mhomed=1
>
> request_route {
>
>
> # per request initial checks
> route(REQINIT);
>
> # NAT detection
> #route(NATDETECT);
>
> # CANCEL processing
> if (is_method("CANCEL")) {
> if (t_check_trans()) {
> route(RELAY);
> }
> exit;
> }
>
> # handle requests within SIP dialogs
> route(WITHINDLG);
>
> ### only initial requests (no To tag)
>
> # handle retransmissions
> if(t_precheck_trans()) {
> t_check_trans();
> exit;
> }
> t_check_trans();
>
> # authentication
> route(AUTH);
>
> # record routing for dialog forming requests (in case they are
> routed)
> # - remove preloaded route headers
> remove_hf("Route");
> if (is_method("INVITE|SUBSCRIBE")) {
> record_route();
> }
>
>
> Thanks in advance.
>
> Diego.
>
>
Hi!
Issue I can’t figure out. Or all working ok and that’s just me who not understands.
I have situation
softPhone (111.222.3.2) -> Kamailio w. rtpproxy (111.222.3.3) -> PBX (111.222.3.4)
All addresses are on same public network.
rtpproxy is running with -l 111.222.3.3 -A 111.222.3.3
I want all media goes through rtpproxy. So, in INVITE and ACK I call rtpproxy_manage().
But according to sip packets on PBX (10.0.0.4) I got media address 10.0.0.2 in SDP, but not 10.0.0.3
If I use scheme, where client in some other NATted network (like 192.168.1.100), all is replaced correctly (means rtpproxy_manage() is working)
So, I can’t understand, rtpproxy_manage() changes only rfc1918 addresses or I’m missing something?
If any - can provide pcap’s with both cases.
Regards, Igor
Hi list,
We have two eth, one with a private ip and the other with a public ip. We
have MHOMED configured. The call comes from private network and then is
routed to a public network via rtjson routing. The is routed but the thing
is that the record route is being set with the private one (It's the same
when calls come from public and is routed to a private network, record
route has the public and not the private).
With mhomed configured shouldn't Record Route be updated as VIA header
".... When activated, sip-router will select a socket that can reach the
destination (to be able to connect to the remote address). (sip-router
opens a UDP socket to the destination, then it retrieves the local IP which
was assigned by the operating system to the new UDP socket. Then this
socket will be closed and the retrieved IP address will be used as IP
address in the Via/Record-Route headers)..."
This is part of ngrep
U 2016/12/22 11:24:15.019643 172.16.213.21:5060 -> 172.16.213.38:5060
INVITE sip:11111111111@172.16.213.38:5060 SIP/2.0.
Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-13024-1-0.
From: "Test" <sip:2222222222@172.16.213.21:5060;user=phone>;tag=1.
To: <sip:1111111111@172.16.213.38:5060>.
Call-ID: 1-13024(a)172.16.213.21.
CSeq: 1 INVITE.
Contact: "2222222222 <sip:2222222222@172.16.213.21:5060>.
....
U 2016/12/22 11:24:15.023923 172.16.213.38:5060 -> 172.16.213.21:5060
SIP/2.0 100 trying -- your call is important to us.
Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-13024-1-0.
....
U 2016/12/22 11:24:15.038695 XXX.XXX.XXX.01:5060 -> XXX.XXX.XXX.02:5060
INVITE sip:1111111111@XXX.XXX.XXX.02:5060 SIP/2.0.
*-----> Record-Route: <sip:172.16.213.38;lr;ftag=1;did=1111>. <----*
Via: SIP/2.0/UDP XXX.XXX.XXX.01;branch=z9hG4bKc09f
Via: SIP/2.0/UDP 127.0.0.8;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.NLOBF6
This is my cfg
mhomed=1
request_route {
# per request initial checks
route(REQINIT);
# NAT detection
#route(NATDETECT);
# CANCEL processing
if (is_method("CANCEL")) {
if (t_check_trans()) {
route(RELAY);
}
exit;
}
# handle requests within SIP dialogs
route(WITHINDLG);
### only initial requests (no To tag)
# handle retransmissions
if(t_precheck_trans()) {
t_check_trans();
exit;
}
t_check_trans();
# authentication
route(AUTH);
# record routing for dialog forming requests (in case they are
routed)
# - remove preloaded route headers
remove_hf("Route");
if (is_method("INVITE|SUBSCRIBE")) {
record_route();
}
Thanks in advance.
Diego.
