Hello Juha:

Thanks a lot for your time and quick response. At this stage, I am interested in the last point (for a bare-bone sort of showcasing) Do I still need a full fledged application (SEMS, JS etc?)

Put another way, can a Kamailio Box & a smartphone (Browser or Linphone) do the job? Or we must have a web page (Apache2) for the show to run (Browser to Browser or APP to Browser A/V call)?

Thanks again!

BR,

Zaka

On 4 July 2016 at 16:15, Juha Heinanen jh@tutpro.com wrote:

Zaka writes:

...

2: Customer will use web browser (websocket compatible of course) and

enter

...

address of the KAMAILIO Box or we shall need another KAMAILIO to statelessly forward the call to next available operator using Hunt Group

or

...

similar scheme?

kamailio could route the call to a call center application, which can easily be written in sems, for example.

...

4: Last, but not the least, how will the users send request for login

from

...

browser? (assuming they have accounts created in the DB on Kamailio Box running WEBSOCKET supported KAMAILIO)

if the user needs to login to your web page, then the websocket sip application can use the same credentials for making the call via kamailio.

-- juha

---

SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Dear List & Dimitry:

Please see the following & advise:

1: I have not been able to make Kamailio Listen on WEBTC port (in my case 8000)

Please see the websocket.cfg DEFs section (with TLS/MSRP disabled)

#!substdef "!DBURL!mysql://kamailio:kamailiorw@localhost/kamailio!g" #!substdef "!MY_IP_ADDR!10.42.0.1!g" #!substdef "!MY_DOMAIN!callcntr.com.al!g" #!substdef "!MY_WS_PORT!8000!g" #!substdef "!MY_WSS_PORT!443!g" #!substdef "!MY_MSRP_PORT!9000!g" #!substdef "!MY_WS_ADDR!tcp:MY_IP_ADDR:MY_WS_PORT!g" ##!substdef "!MY_WSS_ADDR!tls:MY_IP_ADDR:MY_WSS_PORT!g" ##!substdef "!MY_MSRP_ADDR!tls:MY_IP_ADDR:MY_MSRP_PORT!g" ##!substdef "!MSRP_MIN_EXPIRES!1800!g" ##!substdef "!MSRP_MAX_EXPIRES!3600!g"

##!define LOCAL_TEST_RUN ##!define WITH_TLS #!define WITH_WEBSOCKETS ##!define WITH_MSRP

and the output of command

*root@callcntr:/usr/local/etc/kamailio# kamailio -Ee -l 10.42.0.1 -dd -cf websocket.cfg0(10634) INFO: <core> [main.c:1911]: main(): private (per process) memory: 8388608 bytes 0(10634) INFO: <core> [ppcfg.c:82]: pp_subst_add(): ### added subst expression: !DBURL!mysql://kamailio:kamailiorw@localhost/kamailio!g 0(10634) INFO: <core> [ppcfg.c:82]: pp_subst_add(): ### added subst expression: !MY_IP_ADDR!10.42.0.1!g 0(10634) INFO: <core> [ppcfg.c:82]: pp_subst_add(): ### added subst expression: !MY_DOMAIN!callcntr.com.al http://callcntr.com.al!g 0(10634) INFO: <core> [ppcfg.c:82]: pp_subst_add(): ### added subst expression: !MY_WS_PORT!8000!g 0(10634) INFO: <core> [ppcfg.c:82]: pp_subst_add(): ### added subst expression: !MY_WSS_PORT!443!g 0(10634) INFO: <core> [ppcfg.c:82]: pp_subst_add(): ### added subst expression: !MY_MSRP_PORT!9000!g 0(10634) INFO: <core> [ppcfg.c:82]: pp_subst_add(): ### added subst expression: !MY_WS_ADDR!tcp:10.42.0.1:8000!gloading modules under config path: /usr/local/lib64/kamailio/modules/ 0(10634) INFO: <core> [sctp_core.c:74]: sctp_core_check_support(): SCTP API not enabled - if you want to use it, load sctp moduleListening on udp: 10.42.0.1:5060 http://10.42.0.1:5060 tcp: 10.42.0.1:5060 http://10.42.0.1:5060Aliases: tcp: callcntr.com.al:5060 http://callcntr.com.al:5060 udp: callcntr.com.al:5060 http://callcntr.com.al:5060* Besides, when it is run without -c flag, it periodically outputs WS_CONN status with [null] members.

netstat confirms that it is only listening on port 5060.

What I am missing?

KR,

Zaka

On 5 July 2016 at 08:49, Nagorny, Dimitry dimitry.nagorny@robot5.de wrote:

Hi Zaka,

I know Juha already answered some of your questions, but I simply wanted to add some additional information regarding your third question.

Yes disabling would do the job, but if you plan to let your customers use Chrome here is some info I ran into using SIP over WebSocket with WebRTC-Clients (most likely you know this already):

•      Chrome does not allow to use WebRTC on unsecure channels. As a
  

result if you switch to SSL (for your WebRTC-Client Page) Chrome is blocking normal Websocket communication so then I had to turn on WSS in Kamailio which included a TLS setup.

Best Regards

*Dimitry Nagorny*

Trainee

*Von:* sr-users [mailto:sr-users-bounces@lists.sip-router.org] *Im Auftrag von *Zaka *Gesendet:* Montag, 4. Juli 2016 15:07 *An:* Kamailio (SER) - Users Mailing List sr-users@lists.sip-router.org *Betreff:* [SR-Users] Kamailio WEBRTC questions/ confusions

Dear List:

I wonder if this has been asked before. Please ignore or point to the appropriate source, if needed.

Simply, I have little experience with SIP servlets implementing WEBRTC/ Click2Call etc...

Considering a Call Centre scenario offering a WEBRTC solution. Following are the confusions regarding requirements:

1: A Kamailio Box (running WEBSOCKET.config)

2: Customer will use web browser (websocket compatible of course) and enter address of the KAMAILIO Box or we shall need another KAMAILIO to statelessly forward the call to next available operator using Hunt Group or similar scheme?

3: If we don't plan to implement TLS/ MSRP (disabling these in the config file should be suffice?)

4: Last, but not the least, how will the users send request for login from browser? (assuming they have accounts created in the DB on Kamailio Box running WEBSOCKET supported KAMAILIO)

With anticipatory thanks & regards,

Zaka

---

SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users