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Re: [Linphone-developers] [Release] Launch of Flexisip proxy server 1.0


From: Peter Villeneuve
Subject: Re: [Linphone-developers] [Release] Launch of Flexisip proxy server 1.0
Date: Mon, 25 May 2015 12:47:44 +0100

Hi Simon,

Thanks for clearing that up. So it seems that flexisip's behaviour is very similar to the rtpengine module in kamailio.
I'm going to try out flexisip and compare the two to see which offers better performance and stability.

Where's the code located? I can't seem to find at https://github.com/BelledonneCommunications

Cheers,
Peter

On Sun, May 24, 2015 at 3:42 PM, Simon MORLAT <address@hidden> wrote:
Hi Peter,

No your understanding is not correct.
If the client puts ICE candidates in the SDP, then Flexisip will add a relay ICE candidate.
Then the two clients (if both support ICE) are free to use the relay candidates from flexisip or not.
In Linphone's implementation, relay is used at first, and if ICE handshake succeeds with a more direct path, the clients rapidly switch (within a few seconds) to the direct path.

If one of the client doesn't support ICE, then relay is used all the time, because there is absolutely no way for the proxy to determine whether the two clients will need it or not.

Best regards,

Simon

2015-05-22 19:08 GMT+02:00 Peter Villeneuve <address@hidden>:
Hi Simon,

That is great news indeed, and thanks for releasing the code. As a longtime kamailio user, I'm curious to see how flexisip handles ICE and media relaying.
From what I already gathered by reading the docs in the wiki, it seems the mediarelay module is an all or nothing proposition. In other words, you can either force all media to be relayed or not, which is less than ideal. Is my understanding correct?

Cheers,
Peter

On Fri, May 22, 2015 at 3:37 PM, Simon MORLAT <address@hidden> wrote:
Hi Developers,

Today we are officially launching and releasing to the open-source our Flexisip SIP proxy server.
Flexisip is a SIP proxy server implementation compliant to RFC3261, written in C++ 11.

The free linphone.org SIP service is powered with Flexisip since 2011.It  enables Linphone users to have their SIP addresses in order to communicate together.

It has the following main features:
  • transports: SIP/UDP, SIP/TCP and SIP/TLS
  • nat aware with built-in media relay module and stun server
  • digest authentication based on external SQL password database or static password file
  • registrar
  • routing based on registrar database or static route file, with forking
  • interconnected with push notifications systems for reliably notifying mobile apps of incoming calls or messages
  • high level event logging for activity monitoring
  • high availability and cluster mode operation for large deployments
Thanks to its modular architecture and its limited number of required dependencies, Flexisip can also perfectly run on small hardwares (embedded systems).

Flexisip is licensed under Affeiro GPLv3 open source license.

Until a new mailing list dedicated to Flexisip is finally setup, this linphone-developers mailing list can be used to submit questions, remarks, patches about the project.

More information is available from http://www.linphone.org/technical-corner/flexisip/overview .

Best regards,

Simon

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