[SR-Users] Flow Diagram for WebRTC Client1 => WebRTC Client2 (via Kamailio and Asterisk)

Fri Dec 15 22:32:50 CET 2017

Can you send over some captures for test #2?

Well, I’d need the private key to see anything, is this a lab env where you
can mess around? If so, share the capture + private key of the certificate
used for wss pls. If you cannot, please share the signaling in plain text
(you can hide any sensible info, just don’t alter the formats)

Is by any chance the call dropping around 30-32secs after answering?

Is NAT involved anywhere in the test?

On Fri, Dec 15, 2017 at 12:44 Wilkins, Steve <swwilkins at mitre.org> wrote:

> 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 at lists.kamailio.org] *On Behalf
> Of *Joel Serrano
>
> *Sent:* Friday, December 15, 2017 2:22 PM
>
>
> *To:* Kamailio (SER) - Users Mailing List <sr-users at 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/${EXTEN},30,t)
>
>
>
> That way, you are sending the call to Kamailio instead of trying to keep
> it local.
>
>
>
> Sorry for the confusion, let me know if you have any doubts.
>
>
>
> Joel.
>
>
>
>
>
>
>
>
>
>
>
> On Fri, Dec 15, 2017 at 10:43 AM, Joel Serrano <joel at gogii.net> wrote:
>
> Hi Steve,
>
>
>
> You mentioned that Kamailio was handling the registration of the clients...
>
>
>
> Therefor, you want to do something like this if you want to be able to
> bridge calls:
>
>
> exten => _900XX,1,Answer()
>         same => n,DumpChan()
>         same => n,Dial(SIP/${EXTEN}@<<kamailio-peer>>,30,t)
>         same => n,HangUp()
>
>
>
> Another valid format:
>
>
>
> exten => _900XX,1,Answer()
>         same => n,DumpChan()
>         same => n,Dial(SIP/<<kamailio-peer>>/${EXTEN},30,t)
>         same => n,HangUp()
>
>
>
>
>
> Where "<<kamailio-peer>>" == the name of the peer you configured in
> sip.conf for Kamailio.
>
>
>
> At the end, the key here is that you are calling SIP/XXXX which is not
> local, if you call PJSIP/XXXX (per your example) you are trying to reach a
> local endpoint (in this case, the device that would have to be directly
> registered with Asterisk).
>
>
>
> Give a try and let me know how it goes.
>
>
>
>
>
> Cheers,
>
> Joel.
>
>
>
> On Thu, Dec 14, 2017 at 8:59 AM, Wilkins, Steve <swwilkins at mitre.org>
> wrote:
>
> Hello,
>
>
>
> Here is my extentions.conf
>
> exten => _900XX,1,Answer()
>
>         same => n,DumpChan()
>
>         same => n,Dial(PJSIP/${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 at lists.kamailio.org] *On Behalf
> Of *Joel Serrano
> *Sent:* Thursday, December 14, 2017 9:29 AM
> *To:* Kamailio (SER) - Users Mailing List <sr-users at 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 at 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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>
>
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