[pjsip] Fwd: My application with sip

Bill Gardner billg at wavearts.com
Thu May 22 09:32:25 EDT 2014


You need a SIP registrar. It will handle registrations and proxy 
INVITEs. RTP will be directed to the other endpoint without passing 
through registrar. If a media relay is needed for NAT traversal you can 
use a TURN relay and enable ICE protocol.

Bill

On 5/21/2014 8:41 PM, ?? wrote:
> Thanks for your reply.
> My application is required in a  Video surveillance system.  In this 
> system, the  signaling layer interaction must use SIP.  So my 
> application is not just a SIP registrar. It should handle the 
> registrations from clients(IP Camera or User Client), then it also can 
> accept and forward the INVITE requests between different clients. 
> Moreover ,the application should forward the video/audio media 
> streaming, but it doesn't need to code/decode, it can only forward the 
> RTP packets.
> For more details:
> -The system topology hiding for  private networks.
> - I need a subscriber database to store the clients information. When 
> clients register to my application, it should verify password stored 
> in database. When recieve INVITE requests, the application can forward 
> between online clients.
> -This specific call flow is a  requirement in my video surveillance 
> system, it's not a specialized telecommunications application. And  my 
> application should fullfill the Performance Testing:
> 1000 Registers per second, 1000 Invite Requests per second, it also 
> needs to  forwards the media streaming until bandwidth comes to 50% of 
> the maximum.
> Thanks,
> Bob
>
>
> 2014-05-19 3:44 GMT+08:00 Luke B <eat.lemons at gmail.com 
> <mailto:eat.lemons at gmail.com>>:
>
>     More detail on what your trying to accomplish would be useful;
>
>     - Do you require b2bua for the network topology? UA's registering
>     behind NAT for example or Topology hiding for public/private
>     networks etc..
>     - Are you looking to create a SIP application server with a
>     subscriber database?
>     - Is this specific call flow a requirement to test client
>     interworking?
>
>     As bill said, the above could be created with Kamailio/OpenSER or
>     Asterisk/Freeswitch to handle registrations.
>
>     If what you're trying to accomplish is a SIP registrar with one of
>     the above, it's outside of the scope of this mailing - feel free
>     to email me directly if you after any more info on that subject/
>
>     Cheers,
>
>     Luke.
>
>
>     On 18 May 2014 16:26, Bill Gardner <billg at wavearts.com
>     <mailto:billg at wavearts.com>> wrote:
>
>         What you have diagrammed is a SIP registrar, for that you can
>         use Kamailio. If you want a B2BUA you can build that easily
>         with pjsip but it would not handle registration messages.
>
>         Bill
>
>
>         On 5/18/2014 9:50 AM, ?? wrote:
>>
>>         Hi, everyone.
>>         I'm working with sip.
>>         I need to realize the following:
>>         I think it will be a B2BUA or may be not.
>>         If I develop the application using PJSIP, how can I begin
>>         quickly?
>>         Please give me some suggestions.
>>         Thanks a lot,
>>         Bob.
>>
>>
>>
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