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.
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 On Behalf Of Wilkins, Steve Sent: onsdag 9. mai 2018 19:15 To: Kamailio (SER) - Users Mailing List 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.
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@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.orgmailto: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.orgmailto: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.
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.
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@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@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@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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
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@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@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
-
Alex Balashov -
Pan Christensen -
Wilkins, Steve