Displaying 2 results from an estimated 2 matches for "3a771".
Did you mean:
32771
2014 Dec 05
0
Inbound call from sip peer to internal webrtc peer fails while internal sip-webrtc calls work
...9hG4bK-d8754z-bd00e9fd46368417-1---d8754z-
> Max-Forwards: 69
> Contact: <sip:771 at AST.ER.ISK.IP:38699;transport=UDP>
> To: <sip:660 at testers.com <mailto:sip%3A660 at testers.com>;transport=UDP>
> From: "771"<sip:771 at testers.com
> <mailto:sip%3A771 at testers.com>;transport=UDP>;tag=41030177
> Call-ID: YWYwMjMwMmZlODEwM2MwODdjZWJmYjc2NjM5ZmIyNzk.
> CSeq: 2 INVITE
> Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS,
> INFO, SUBSCRIBE
> Content-Type: application/sdp
> Supported: replaces, norefersub, exte...
2014 Dec 05
2
Inbound call from sip peer to internal webrtc peer fails while internal sip-webrtc calls work
Hello,
I'd appreciate your comments on the following problem I'm having, please
forgive me if this is something obvious, I've been scratching my head on
this for a while:
I have Asterisk+Kamailio setup where I'm currently testing inbound calls
from outside. I have both webrtc and sip clients, where webrtc peers are
defined according to sip.js instructions (