ernal IP via a start
parameter. Is that possible?
Any help would be grateful.
Kind regards,
Marko Seidenglanz
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The question is, how I can forward the BYE message back to asterisk?
2014-10-29 15:24 GMT+01:00 Marko Seidenglanz :
> Thank you for the fast reply,
>
> I have enabled NAT Traversal like in the default config. The problem seems
> to be that the cannot be assigned to any transaction.
> Daniel
>
>
> On 29/10/14 11:55, Marko Seidenglanz wrote:
>
> Hello,
>
> We have a setup where Kamailio 4.2 is used in front of Asterisk as
> WebRTC Proxy doing the encryption and NAT Traversal.
>
> Everything works as expected, except that BYE Requests sent by the
Hello,
We have a setup where Kamailio 4.2 is used in front of Asterisk as WebRTC
Proxy doing the encryption and NAT Traversal.
Everything works as expected, except that BYE Requests sent by the WebRTC
Client are not forwarded by Kamailio to Asterisk. We use record routing.
Instead Kamailio respon
Hello,
We want to use Kamailio (4.2) + RTPEngine (3.3) behind NAT. Unfortunately
the STUN Messages (Binding Requests) sent from remote peer (Google Chrome )
do not arrive at RTPEngine host.
I think they get blocked by the firewall. RTPEngine itself does not send
any STUN Messages until it get's t
1 GMT+02:00 Marko Seidenglanz :
> Hello,
>
> We are trying to use Kamailio 4.2 with RTPEngine 3.3 behind NAT.
>
> Somehow SIP responses (200 OK) are not handled by Kamailio.
>
>
>
> The following INVITE is send to the receiving end (WebRTC Client):
>
> INVITE sip:ev
Hello Camille,
thank you for your fast response.
I have defined a reply route:
onreply_route[REPLY_FROM_WEBRTC] {
xlog("L_DEBUG", "<<<
SIP WEBRTC_REPLY: $si:$sp --> $du ");
xlog("L_DEBUG", "WEBRTC
Hello,
We are trying to use Kamailio 4.2 with RTPEngine 3.3 behind NAT.
Somehow SIP responses (200 OK) are not handled by Kamailio.
The following INVITE is send to the receiving end (WebRTC Client):
INVITE sip:evqzk4l62mdf3g7u6e...@whtest3.24dial.com SIP/2.0
Record-Route:
Via: SIP/2.0/UDP 10
Hello,
I have a problem with the following configuration.
I want to make calls from Asterisk to a browser using RTPEngine as relay.
Everything works fine, if Kamailio is not natted (See
kamailio_without_nat.log).
If it's address is translated, then 200 OK responses from the browser don't
seem t
Hello,
I have a problem with the following configuration.
I want to make calls from Asterisk to a browser using RTPEngine as relay.
Everything works fine, if Kamailio is not natted (See
kamailio_without_nat.log).
If it's address is translated, then 200 OK responses from the browser don't
seem t
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