[asterisk-dev] 13 November, 2012 New SIP stack update

[Daniel Pocock]

On 14/11/12 15:16, Olle E. Johansson wrote:


> Unfortunately that's what I see with both PJsip, Resiprocate and Sofia users - many private "forks". 

I'd be interested to see examples of the private forks of resiprocate

There are many branches in SVN, but most of them started with the
intention to merge back into trunk.

> I have no insight into why this is the state of the SIP stack "industry". The discussion I've had with

Some people are developing for Windows, mobile or other platforms.
Their packaging needs are very different.

E.g. on Windows it is common to embed libraries and link statically.

On mobile, there is no dependency-based packaging system, so projects
like Lumicall (mjsip) and CSipSimple (pjsip) just embed a (forked) SIP stack


> 
> I would like to see another discussion soon - how we should change the core PBX in order
> to be able to build a proper SIP stack. If Digium and the community invest in this development,
> I would like to see an upgrade of the core so that we can do things right. In my installations,
> I have almost only SIP. I do believe a majority of Asterisk channels today are SIP. 

One other possibility: maybe drop things like TLS support and encourage
people to use a proxy for that.  Obviously Asterisk will not know what
was in the certs, so it won't be able to pass such details through to
the dialplan environment, but otherwise it may be technically sound and
easier to support.