[asterisk-dev] internal_timing on 1.2? clock drift?

[Daniel Pocock]

Kevin P. Fleming wrote:

>----- Daniel Pocock <daniel at readytechnology.co.uk> wrote:
>  
>
>>Is the internal_timing patch likely to be backported into a future
>>1.2.x 
>>release, or do people just have to wait for 1.4?
>>    
>>
>
>We do not add new functionality to release branches, except in cases where it is necessary to solve security/DoS vulnerabilities.
>
>  
>
I would consider this to be more of a bug fix than a new feature - with 
systems like ENUM now starting to become more common, people will 
potentially use their Asterisk to communicate with a multitude of remote 
peers, not just one or two SIP providers.  It simply can't be assumed 
that every one of those remote peers will have silence suppression disabled.

Quite a few people have had a bad encounter with silence suppression 
within their first week of using Asterisk, depending upon their SIP 
gateway, their handsets, and how closely they had read the 
documentation.  Some have probably just given up, or decided to wait 
another 6 months before having another go.  Surely that isn't good 
publicity for Digium's flagship product?

The more conservative Asterisk deployer will probably refrain from 
enabling ENUM lookups until this issue is resolved - a sad situation 
when you consider the fact that we should be trying to push ENUM and SIP 
as the superior, open alternatives to a proprietary VoIP network that is 
quickly becoming the defacto standard for the consumer.