Asterisk 13 +webrtc2sip + sipml5

[Bleddyn Raw-Rees]

Hello community! I was wondering whether you could try and help me get my head around the problem that I have? I'm very new to these technologies but I've been stumbling my way through (with extensive googling) until this point, where I've hit a slight wall. I'll briefly give you the background and then pose the problem. 

In order to test webrtc2sip as webRTC/SIP gateway, I've compiled it on a virtual machine (centos) - following the webrtc2sip techincal guide by Mamadou DIOP [http://webrtc2sip.org/].

I've then built a second centos VM running asterisk 13. This is my SIP server. I have a third VM running windows with a softphone that I can use to register two endpoints to the SIP server. I can make a call between both endpoints (on the same softphone or on a second softphone on a fourth VM) . The asterisk configuration is very basic, just enough to get that phone call to work. I've only edited two config files. i.e. pjsip.conf and extensions.conf and have done so following an example in the asterisk wiki [https://wiki.asterisk.org/wiki/display/AST/Basic+PBX+Functionality]

pjsip.conf:

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[transport-udp]

type=transport

protocol=udp

bind=0.0.0.0

;Templates for the necessary config sections

[endpoint_internal](!)

type=endpoint

context=from-internal

disallow=all

allow=ulaw

[auth_userpass](!)

type=auth

auth_type=userpass

[aor_dynamic](!)

type=aor

max_contacts=1

;Definitions for our phones, using the templates above

[demo-alice](endpoint_internal)

auth=demo-alice

aors=demo-alice

[demo-alice](auth_userpass)

password=unsecurepassword ; put a strong, unique password here instead

username=demo-alice

[demo-alice](aor_dynamic) 

[demo-bob](endpoint_internal)

auth=demo-bob

aors=demo-bob

[demo-bob](auth_userpass)

password=unsecurepassword ; put a strong, unique password here instead

username=demo-bob

[demo-bob](aor_dynamic) 

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AND extensions.conf:

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[from-internal]

exten=>6001,1,Dial(PJSIP/demo-alice,20)

exten=>6002,1,Dial(PJSIP/demo-bob,20)

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NOW THEN..... this is the bit that I'm a bit stumped by!

I now want to use sipml5 to make a call to my softephone that is registered to asterisk, but this isn't working. The VMs are connected on the same VLAN as follows:

[sipml5]{192.168.254.130}<------>[webrtc2sip]{192.168.254.128}<------>[asterisk SIP server]{192.168.254.129}

The first question should probably be: 1) should I be able to register an endpoint to the SIP server from the web browser at this point?

2) How is the asterisk server supposed to see the web browser(chrome) running sipml5 if its via another server?

3) Do I need to some how set the SIP servers IP address as the gateway of the webrtc2sip server? Or maybe configure some sort of trunk?

4) Or alternatively try and configure asterisk such that registration isn't necessary?

5) I'm also looking at the config.xml file of webrtc2sip and there are some example SIP addresses that I'm sure need changing..... could it be as simple as editing this file?

I understand that there's quite a lot here to answer so I would be incredibly grateful just for a line of inquiry!