Thanks Richard.
It worked.
*Thanks & Regards,*
Amit
On 2/14/2015 7:02 PM, Richard Fuchs wrote:
On 14/02/15 08:13 AM, Amit Patkar wrote:
Hi
I am getting error with rtpengine.
Running Kamailio 4.2.3
I am trying to call from conventional SIP client to WebRTC client
Google Chrome v38.0.2125.10
On 14/02/15 08:13 AM, Amit Patkar wrote:
> Hi
>
> I am getting error with rtpengine.
> Running Kamailio 4.2.3
>
> I am trying to call from conventional SIP client to WebRTC client
>
> Google Chrome v38.0.2125.104
> Firefox v33.0
>
> Using sipML5 as WebRTC client
>
> root@rtcpbx:/home/avhan# /u
Hi
I am getting error with rtpengine.
Running Kamailio 4.2.3
I am trying to call from conventional SIP client to WebRTC client
Google Chrome v38.0.2125.104
Firefox v33.0
Using sipML5 as WebRTC client
root@rtcpbx:/home/avhan# /usr/sbin/rtpengine
--interface=127.0.0.1\!192.168.2.161 --listen-n
On 13 Feb 2015, at 20:07, Richard Fuchs wrote:
> Load balancing is achieved by running a hash over the call-id and using
> the hash value to determine which RTP proxy from the selected set to
> use. The hash ensures that everything related to the same call ends up
> on the same RTP proxy, which