[asterisk-dev] classifying SIP peers

Olle E. Johansson oej at edvina.net
Tue Feb 11 13:18:54 CST 2014

On 11 Feb 2014, at 19:56, Matthew Jordan <mjordan at digium.com> wrote:

> "Stable" or "production" are things that you have to determine for
> yourself. I don't know your system, or your deployment requirements.
> It's open source software; if it works for your system, then great. If
> not, then the good news is that it's open source! There's a community
> of developers around to help you figure out what is going on.

> Speaking only for Digium, I will point out that we are actively
> developing, improving, and testing the PJSIP stack in Asterisk. Going
> forward, I would expect that - barring bug fixes in chan_sip - the new
> SIP stack is what will receive the vast majority of our attention.

While I understand Digium will focus on that, I am sure that many of us will
still contribute to chan_sip as long as that is in production and
has features that pjsip does not have.

I have a lot of patches for chan_sip around that needs moving forward,
but many more for the RTP layer.


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