[asterisk-dev] SIP TCP/TLS, release policy and more (personal opinions included)

[Johansson Olle E]

20 okt 2008 kl. 18.04 skrev Kevin P. Fleming:

> Johansson Olle E wrote:
>
>> I think we're in disagreement there. As you know, I've been heavily
>> involved
>> with the SIP channel during many years. It is important for me to
>> distance
>> myself from this work that I do consider as broken. Even though UDP
>> mostly
>> works, the changes are so big inside the SIP channel that I'm still  
>> not
>> convinced that it is up to par with the 1.4 stack. As long as I am  
>> not
>> sure
>> of that, I do reserve the right to consider the stack broken. I do  
>> know
>> that you have another opinion, and respect you for that.
>
> Also, the changes that have gone into chan_sip *after* 1.6.0 are much
> larger and much more invasive than the TCP/TLS changes were, and they
> directly affect the existing UDP support. We need to be very careful
> here when saying that the TCP/TLS code might be the reason for the
> current known (and unknown issues), when the testing was done using  
> code
> that has moved far beyond the 1.6.0 release already.

I did in fact point at my kill-the-user and Murf's changes in my  
original mail :-)

/O