[Asterisk-Dev] Developers - How involved, and what cost? (SIP Jitter Buffer)

[Phil Doroff]

Alright..

 From my (rather limited!) understanding, there is currently no jitter buffer
support (or even the beginnings of such) in asterisk.

Due to some company requirements, we're looking for some cost estimates on
contracting a current asterisk developer to at least introduce a
beta-quality (but usable in production) SIP jitter buffer to the current
source tree.

Requirements:

1. First and foremost, any development done must be re-introduced back into
the source tree for public use.  It's your responsibility to ensure you are
not under any previous contractual (employment or otherwise) obligations
that might hinder this.

2. Once we figure out what a reasonable cost for this is, we will pay on a
per-job basis.  If this doesn't interest you I'm sorry, but it's all we can
offer currently.

3.  Be willing to take over testing/development of this.  We can send some
hardware to use as a test platform, but other than that we are most
assuredly not asterisk or PBX experts by any means. :)

So.. comments welcome.  Let me know if you're interested in the job, and how
involved you think the process will be.  The goal is to have a 20-30ms
average jitter connection send understandable voice quality through.
Currently, outbound jitter makes our phone system almost unusable for some
employees.  Our experience hiring a developer off this list to fix a Cisco
79xx series phone DTMF issue were positive, so we'd like to continue this.

Thanks much,

Phil Doroff
Reflected Networks, Inc.
phil at reflected.net