15 Dec
2017
15 Dec
'17
11:43 p.m.
Hi Joel,
I have tried this along with setting outbound_proxy. Client2 sees the call from Client1,
I click on “Answer” and the call eventually just drops, although the camera showing myself
does come on after I click on “Answer”. Others on this board have given it a real try to
help, but I fear I am missing some deep concept. Note: I have looked at tcpdumps and
asterisk logs for a few weeks now!
Just for grins, here are my test cases.
For the sake of making things a little easier to read I will define –
Asterisk Server: AS
Kamailio Server KS
WebRTC Client1 and Client2 are WebRTC (tryit) instances both using chrome
Test1 (tryit configured to go to AS via wss): Client1 can call Client2; Client2 answers
and I can see Audio/Video on both Client1 and Client2 (Success!);
The tcpdump (AS) shows no SIP, RTP, SRTP activity, but a lot of UDP activity.
Test2 (tryit configured to go KS via wss): Client1 calls Client2, Client2 rings and show’s
“ANSWER” and after I answer Client1 hangs up (Failure!)
Test3 (tryit configured to go KS via wss): Client1 calls an Asterisk Dial-Plan which plays
audio and video (Success!)
From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf Of Joel Serrano
Sent: Friday, December 15, 2017 2:22 PM
To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org>
Subject: Re: [SR-Users] Flow Diagram for WebRTC Client1 => WebRTC Client2 (via Kamailio
and Asterisk)
Hi Steve,
Sorry, I was replying at the same time in another thread and I mixed responses.
Let me correct myself:
You are using:
same => n,Dial(PJSIP/${EXTEN},30,t)
So if someone dials "client1" you would try to reach PJSIP/client1 and that
would fail because client1 is not registered with Asterisk.
Let's say your trunk with Kamailio si called "kamailio" in pjsip.conf...
If "client1" is registered with Kamailio instead of Asterisk, you want to send
that call to Kamailio:
same => n,Dial(PJSIP/${EXTEN}@kamailio-trunk,30,t)
same => n,Dial(PJSIP/kamailio-trunk/${EXTEN},30,t)
That way, you are sending the call to Kamailio instead of trying to keep it local.
Sorry for the confusion, let me know if you have any doubts.
Joel.
On Fri, Dec 15, 2017 at 10:43 AM, Joel Serrano
<joel@gogii.net<mailto:joel@gogii.net>> wrote:
Hi Steve,
You mentioned that Kamailio was handling the registration of the clients...
Therefor, you want to do something like this if you want to be able to bridge calls:
exten => _900XX,1,Answer()
same => n,DumpChan()
same => n,Dial(SIP/${EXTEN}@<<kamailio-peer>>,30,t)
same => n,HangUp()
Another valid format:
exten => _900XX,1,Answer()
same => n,DumpChan()
same => n,Dial(SIP/<<kamailio-peer>>/${EXTEN},30,t)
same => n,HangUp()
Where "<<kamailio-peer>>" == the name of the peer you configured in
sip.conf for Kamailio.
At the end, the key here is that you are calling SIP/XXXX which is not local, if you call
PJSIP/XXXX (per your example) you are trying to reach a local endpoint (in this case, the
device that would have to be directly registered with Asterisk).
Give a try and let me know how it goes.
Cheers,
Joel.
On Thu, Dec 14, 2017 at 8:59 AM, Wilkins, Steve
<swwilkins@mitre.org<mailto:swwilkins@mitre.org>> wrote:
Hello,
Here is my extentions.conf
exten => _900XX,1,Answer()
same => n,DumpChan()
same => n,Dial(PJSIP/${EXTEN},30,t)
same => n,HangUp()
I have 90001,90002 in pjsip.conf with a webrtc endpoint.
Thank you,
-Steve
From: sr-users
[mailto:sr-users-bounces@lists.kamailio.org<mailto:sr-users-bounces@lists.kamailio.org>]
On Behalf Of Joel Serrano
Sent: Thursday, December 14, 2017 9:29 AM
To: Kamailio (SER) - Users Mailing List
<sr-users@lists.kamailio.org<mailto:sr-users@lists.kamailio.org>>
Subject: Re: [SR-Users] Flow Diagram for WebRTC Client1 => WebRTC Client2 (via Kamailio
and Asterisk)
Hi, can you share with us the asterisk dialplan part where you call the Dial()
application?
On Tue, Dec 12, 2017 at 06:38 Wilkins, Steve
<swwilkins@mitre.org<mailto:swwilkins@mitre.org>> wrote:
Hello All,
I am looking for a Diagram or such that shows the flow of SIP traffic for a WebRTC Client1
=> WebRTC Client2 call using Kamailio in front of Asterisk.
I am unable to get Asterisk to find the correct registered clients, which are registered
in Kamailio and am hoping verifying the flow will help give me a clue as to what is going
on. E.g. Using chrome and tryit-pjsip I have Client1, and Client2 registered in Kamailio.
However when I try to connect Client1 to Client2 (make a call), Asterisk has no clue where
Client1 and Cleint2 are registered to.
Thank you!
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