[SR-Users] Kamailio WEBRTC questions/ confusions

[Zaka]

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 at 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://10.42.0.1:5060>Aliases:              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 at 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 at lists.sip-router.org] *Im
> Auftrag von *Zaka
> *Gesendet:* Montag, 4. Juli 2016 15:07
> *An:* Kamailio (SER) - Users Mailing List <sr-users at 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 at lists.sip-router.org
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>
>


-- 
Best regards,

Zaka Bhatti
Tirana,
Republic of Albania
+355 672 000 771

https://al.linkedin.com/in/kamailian


C2C (Click to Call)

http://click2dial.org/u/emFrYS5iaGF0dGlAZ21haWwuY29t

One accurate measurement is worth more than a thousand expert opinions!
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