Displaying 2 results from an estimated 2 matches for "3a660".
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3660
2014 Dec 05
0
Inbound call from sip peer to internal webrtc peer fails while internal sip-webrtc calls work
On 05/12/14 16:46, Olli Heiskanen wrote:
> INVITE that Asterisk (at port 5070) receives:
> PU.BL.IC.IP:5060 > PU.BL.IC.IP:5070: SIP, length: 1046
> INVITE sip:660 at testers.com
> <mailto:sip%3A660 at testers.com>;transport=UDP SIP/2.0
> Record-Route: <sip:PU.BL.IC.IP;lr=on;ftag=41030177>
> Via: SIP/2.0/UDP
> PU.BL.IC.IP;branch=z9hG4bKd7b.ca8b6ac6a82d605cf658af0fea7c9e86.0
> Via: SIP/2.0/UDP
> AST.ER.ISK.IP:38699;rport=38699;branch=z9hG4bK-d8754z-bd00e9fd46368417-1--...
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 (