9 May
2018
9 May
'18
8:15 p.m.
Hello All,
I am trying to resolve, in my mind, the flow of a WebRTC<=>WebRTC call using
Kamailio and Asterisk.
Each WebRTC client is registered in Kamailio and when I call WebTRC Client1 from WebRTC
Client2 what I see is ->
The Invite is sent from Kamailio to Asterisk and then Asterisk is sending the Invite back
to Kamailio. Also depending on
The version of Asterisk, the INVITE will then get forwarded to the AOR that is registered
in Kamailio for the called number.
Does this seem correct? It seems like there is an extra hop in there.
The reason I am now very curious now is because everything works fine if using Kamailio
5.0 and Asterisk 14.6, but I switch to Asterisk 15.3
I get the extra hop and call is dropped after 30 seconds.
I would appreciate any thoughts on this.
Thank you in advance.
11 May
11 May
5:19 p.m.
New subject: Question about using Kamailio and Asterisk and flow of an "INVITE"
Hello Steve.
What are you trying to achieve?
The call could go from client A to Kamailio to client B. No need to involve Asterisk. If
you need PBX functionality, the INVITE needs to be routed to Asterisk, which will most
likely answer the call and then set up a new call to client B. As Asterisk doesn't
know where client B is, it needs to route this new call to Kamailio where client B is
registered. It's possible for Asterisk to know where client B is but that solves
nothing and may create other problems.
With kind regards
Pan B. Christensen
Developer
Phonect AS
From: sr-users <sr-users-bounces(a)lists.kamailio.org> On Behalf Of Wilkins, Steve
Sent: onsdag 9. mai 2018 19:15
To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org>
Subject: [SR-Users] Question about using Kamailio and Asterisk and flow of an
"INVITE"
Hello All,
I am trying to resolve, in my mind, the flow of a WebRTC<=>WebRTC call using
Kamailio and Asterisk.
Each WebRTC client is registered in Kamailio and when I call WebTRC Client1 from WebRTC
Client2 what I see is ->
The Invite is sent from Kamailio to Asterisk and then Asterisk is sending the Invite back
to Kamailio. Also depending on
The version of Asterisk, the INVITE will then get forwarded to the AOR that is registered
in Kamailio for the called number.
Does this seem correct? It seems like there is an extra hop in there.
The reason I am now very curious now is because everything works fine if using Kamailio
5.0 and Asterisk 14.6, but I switch to Asterisk 15.3
I get the extra hop and call is dropped after 30 seconds.
I would appreciate any thoughts on this.
Thank you in advance.
13 May
13 May
4:14 p.m.
New subject: Question about using Kamailio and Asterisk and flow of an "INVITE"
Hello Pan,
Thank you for responding!,
In my configuration I have to use Asterisk as my PBX, and I use Kamailio in front of
Asterisk accepting and inspecting calls.
I have many AORs, for which a phone can register to. I have noticed that, depending on
the call, when a call arrives it gets REGISTERED in
Kamailio (I do not forward the registration to Asterisk); Kamailio sends the INVITE to
Asterisk, and then Asterisk send the INVITE back to
Kamailio. The calls seem to work fine (Duplex Audi/Video). I think I know why the INVITE
is forwarded to Kamailio but have not been able
to work around it; the AOR's all have a Contact of Kamailio and I think this is the
reason for the forwarding of the INVITE. If I don't set Kamailio as the Contact for
the AOR's, calls do not work.
Thanks Again,
-Steve
From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf Of Pan Christensen
Sent: Friday, May 11, 2018 10:20 AM
To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org>
Subject: Re: [SR-Users] Question about using Kamailio and Asterisk and flow of an
"INVITE"
Hello Steve.
What are you trying to achieve?
The call could go from client A to Kamailio to client B. No need to involve Asterisk. If
you need PBX functionality, the INVITE needs to be routed to Asterisk, which will most
likely answer the call and then set up a new call to client B. As Asterisk doesn't
know where client B is, it needs to route this new call to Kamailio where client B is
registered. It's possible for Asterisk to know where client B is but that solves
nothing and may create other problems.
With kind regards
Pan B. Christensen
Developer
Phonect AS
From: sr-users
<sr-users-bounces@lists.kamailio.org<mailto:sr-users-bounces@lists.kamailio.org>>
On Behalf Of Wilkins, Steve
Sent: onsdag 9. mai 2018 19:15
To: Kamailio (SER) - Users Mailing List
<sr-users@lists.kamailio.org<mailto:sr-users@lists.kamailio.org>>
Subject: [SR-Users] Question about using Kamailio and Asterisk and flow of an
"INVITE"
Hello All,
I am trying to resolve, in my mind, the flow of a WebRTC<=>WebRTC call using
Kamailio and Asterisk.
Each WebRTC client is registered in Kamailio and when I call WebTRC Client1 from WebRTC
Client2 what I see is ->
The Invite is sent from Kamailio to Asterisk and then Asterisk is sending the Invite back
to Kamailio. Also depending on
The version of Asterisk, the INVITE will then get forwarded to the AOR that is registered
in Kamailio for the called number.
Does this seem correct? It seems like there is an extra hop in there.
The reason I am now very curious now is because everything works fine if using Kamailio
5.0 and Asterisk 14.6, but I switch to Asterisk 15.3
I get the extra hop and call is dropped after 30 seconds.
I would appreciate any thoughts on this.
Thank you in advance.
4:19 p.m.
New subject: Question about using Kamailio and Asterisk and flow of an "INVITE"
On Sun, May 13, 2018 at 01:14:08PM +0000, Wilkins, Steve wrote:
In my configuration I have to use Asterisk as my PBX,
and I use
Kamailio in front of Asterisk accepting and inspecting calls. I have
many AORs, for which a phone can register to. I have noticed that,
depending on the call, when a call arrives it gets REGISTERED in
Kamailio (I do not forward the registration to Asterisk); Kamailio
sends the INVITE to Asterisk, and then Asterisk send the INVITE back
to Kamailio. The calls seem to work fine (Duplex Audi/Video). I
think I know why the INVITE is forwarded to Kamailio but have not been
able to work around it; the AOR's all have a Contact of Kamailio and I
think this is the reason for the forwarding of the INVITE. If I don't
set Kamailio as the Contact for the AOR's, calls do not work.
Are you sure that you are using the word AOR correctly?
If you're not forwarding registrations to Asterisk but instead storing
them in Kamailio's registrar, setting the Contact binding of an AOR to
have the address of that very same Kamailio registrar would make no
sense.
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
5:31 p.m.
New subject: Question about using Kamailio and Asterisk and flow of an "INVITE"
Yes, that is what has me a little boggled. If I don't, calls do not go through so I
am obviously confused, but that is how I was finally able to get full duplex calls
working.
In Asterisk, I have a Kamailio endpoint listening for traffic from Kamailio. How would
Asterisk know the Contact information if Asterisk does not re-contact Kamailio for that
Information (since all the registration information is on Kamailio).
I was under the impression that it is best to keep the registrations on Kamailio, why
forward them. Below is my pjsip.conf.
When a user calls 30001(a)192.21.1.5 (Kamailio IP), the call is registered in Kamailio, and
Asterisk can contact Kamailio to get the
Contact information it needs.
Pjsip.conf
[kamailio](!)
type=endpoint
context=from-internal
transport=transport-tcp
media_address=100.20.30.125 //Asterisk PBX IP
...
aors=kamailio
[kamailio](kamailio)
aors=kamailio
[kamailio]
type=aor
contact=sip:192.21.1.5:5060
[kamailio]
type=identify ; Must be of type identify (default: "")
endpoint=kamailio
match=192.21.1.5
[30001](webrtc) //tls endpoint
auth=auth30001
aors=30001
[auth30001](auth-userpass)
password=12345
username=30001
[30001](aor-single-reg)
contact=sip:30001@192.21.1.5:5060
Am I totally wrong, even though it works?
Thank you all very much!
-Steve
-----Original Message-----
From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf Of Alex Balashov
Sent: Sunday, May 13, 2018 9:19 AM
To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org>
Subject: Re: [SR-Users] Question about using Kamailio and Asterisk and flow of an
"INVITE"
On Sun, May 13, 2018 at 01:14:08PM +0000, Wilkins, Steve wrote:
In my configuration I have to use Asterisk as my PBX,
and I use
Kamailio in front of Asterisk accepting and inspecting calls. I have
many AORs, for which a phone can register to. I have noticed that,
depending on the call, when a call arrives it gets REGISTERED in
Kamailio (I do not forward the registration to Asterisk); Kamailio
sends the INVITE to Asterisk, and then Asterisk send the INVITE back
to Kamailio. The calls seem to work fine (Duplex Audi/Video). I
think I know why the INVITE is forwarded to Kamailio but have not been
able to work around it; the AOR's all have a Contact of Kamailio and I
think this is the reason for the forwarding of the INVITE. If I don't
set Kamailio as the Contact for the AOR's, calls do not work.
Are you sure that you are using the word AOR correctly?
If you're not forwarding registrations to Asterisk but instead storing them in
Kamailio's registrar, setting the Contact binding of an AOR to have the address of
that very same Kamailio registrar would make no sense.
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
5:32 p.m.
New subject: Question about using Kamailio and Asterisk and flow of an "INVITE"
That seems correct and sounds like it should work. What is the request
URI of the INVITE that comes back into Kamailio on the B leg? Does its
domain match the AOR domain? Do you have your registrar/usrloc set to
use domains on lookup()?
-- Alex
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
5:59 p.m.
New subject: Question about using Kamailio and Asterisk and flow of an "INVITE"
Hi Alex,
On a WebRTC to WebRTC call (both UAC's registered in Kamailio) =>
- When Asterisk sends the INVITE back to Kamailio, it has the id of the call and the IP
Address of the laptop the call was made from
INVITE
sip:vn768thu@3stum5dgqf9j.invalid;transport=ws;alias=133.148.203.97~53778~6;alias=133.148.203.97~53778~6;ob
SIP/2.0
- I have been using IP Addresses instead of Domains, as I have had much more luck doing it
that way.
Should I switch to FQDN and Domains? If so, may I ask what all would need to be
changed?
Thank you for your time.
-Steve
-----Original Message-----
From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf Of Alex Balashov
Sent: Sunday, May 13, 2018 10:32 AM
To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org>
Subject: Re: [SR-Users] Question about using Kamailio and Asterisk and flow of an
"INVITE"
That seems correct and sounds like it should work. What is the request URI of the INVITE
that comes back into Kamailio on the B leg? Does its domain match the AOR domain? Do you
have your registrar/usrloc set to use domains on lookup()?
-- Alex
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
2389
days inactive
2393
days old
6 comments
3 participants
participants (3)
-
Alex Balashov -
Pan Christensen -
Wilkins, Steve