n given instance sizes.
>
> On March 27, 2017 3:09:56 PM EDT, Jade SZ wrote:
> >Hi Guys,
> >
> >I am running a simple REGISTER load test on:
> >
> >1) Kamailio sever with 2 cores - mem 5G
> >2) Kamailio server with 4 cores - mem 16G
> >
> >Both
Hi Guys,
I am running a simple REGISTER load test on:
1) Kamailio sever with 2 cores - mem 5G
2) Kamailio server with 4 cores - mem 16G
Both are EC2 instances.
At -r = 500 i.e. 500 reg/sec sipP test works fine with very few re-trans.
But when i increase it to 800 reg/sec it starts retransmissio
Hi Guys,
SIPp is great in testing pure SIP udp/tcp/tls but I was wondering if there
is a similar tool for stress testing Kamailio websocket2sip.
I am trying to load-test webrtc2sip kamailio's implementation closest to
the real scenario. One way would be to fork multiple process of webrtc2sip
clie
Anyone from community?
On Wed, Mar 1, 2017 at 11:41 AM, Jade SZ wrote:
> mistakenly pressed sent earlier, here is complete query.
>
> If we want to use multiple kamailio servers for WebRTC based user agents,
> what would be the best practice:
>
> 1) WS/WSS Outbound/Edge Prox
proxy will have its own address, making client
side to be restricted to use one proxy only.
Looking forward to some directions here. Thanks
On Wed, Mar 1, 2017 at 11:25 AM, Jade SZ wrote:
> Hi Guys,
>
> I am using kamailio/rtpengine and webrtc over sip client for a/v calls and
> al
Hi Guys,
I am using kamailio/rtpengine and webrtc over sip client for a/v calls and
all works fine.
Now I want to scale it to 2 or more than 2 kamailio servers. As per search
on mailing list I could:
1) Websocket Outbound/Edge Proxy
2) Each UA client to register with its
Regards,
Jade SZ
Horizontal scaling, that is what I am looking for webrtc over sip. Also by
giving 1 SIP URI to webrtc client.
Regarding - "Why not do UA -> kamailio 1 -> kamailio 2 or kamailio 3?"
Because I want the UA to be configured to wss which can load-balance to
multiple kamailio servers. Avoiding single p
ctual true remote proxy
> endpoint?
>
> On Mon, Feb 27, 2017 at 11:04 PM, Jade SZ wrote:
>
>> Hi Team,
>>
>> Question:
>>
>> Just wanted to clarify regarding SIP URI in webrtc (over sip) connection.
>>
>> e.g. if we have a scenario where Kama
Hi Team,
Question:
Just wanted to clarify regarding SIP URI in webrtc (over sip) connection.
e.g. if we have a scenario where Kamailio is hosted with websocket support.
Websocket URI is used to send packets to to wss/ws address and the SIP URI
goes with it. I have tested it with any SIP URI incl
le us to have on the same 443 port regular
>> Web server and SIP WebSockets, for now, it works pretty well.
>>
>> --
>> Ludovic Gasc (GMLudo)
>> Lead Developer Architect at ALLOcloud
>> https://be.linkedin.com/in/ludovicgasc
>>
>> 2017-02-02 18:39 GMT+01:
Hi Guys,
I am trying to setup the following flow:
Browser >> WSS >> HA Proxy >>> WSS >> Kamailio
But getting TLS errors in Kamailio logs:
*[29634]: ERROR: [tcp_read.c:1321]: tcp_read_req(): ERROR:
tcp_read_req: error reading - c: 0x7f68ebe872b0 r: 0x7f68ebe87330*
*[29631]: ERROR: tls [tls_util.
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