Webrtc Client Support for dialing sip uri

[Raj K]

Does any version of Webrtc Client supports dialing direct sip uri rather than user part of sip uri?

For e.g. using Webrtc client GUI , dialing to "sip:12...@xyz.com" rather than dialing "12345".

Regards

Rajendra

[aaroncray]

I'm not sure if you're asking if there is a software product that does that or you're asking if the client libraries of webrtc support that? So 1) I'm not sure if anyone has made that and for 2) webrtc is independent of signaling (SIP, JSEP, etc...). SIP will do fine for signaling so it's absolutely possible to make that work. Cheers!

-Aaron

On Thursday, February 13, 2014 8:31:58 AM UTC-8, Raj K wrote:

Does any version of Webrtc Client supports dialing direct sip uri rather than user part of sip uri?

For e.g. using Webrtc client GUI , dialing to "sip:...@xyz.com" rather than dialing "12345".

Regards

Rajendra

[Warren McDonald]

Hi Rajendra,

the sample pages for the SIPml5 JS library can be used to dial a fully qualified SIP URI. The interface will use the realm specified in the SIP stack config for short name/id strings entered. If you enter a full URI it will use it. 

Because of media codecs and encryption you will be probably limited to G.711 voice calls unless you use a gateway like webrtc2sip. If the SIP points to an endpoint that will go via PSTN, then you will also need a subscription to a paid SIP service. This can also be configured into the stack so calls are routed through the correct service.

Warren 

On Friday, 14 February 2014 03:31:58 UTC+11, Raj K wrote:

Does any version of Webrtc Client supports dialing direct sip uri rather than user part of sip uri?

For e.g. using Webrtc client GUI , dialing to "sip:...@xyz.com" rather than dialing "12345".

Regards

Rajendra