In an ideal world, yes. In the real world it's difficult enough to get predictable results negotiating ptime across two different Sonus implementations... I'm not beating up on Sonus here either. Bodies like SIPConnect are in existence because of the wide range of capability and interpretation of spec currently present in the VoIP space. I don't want to seem like a pessimist but for the moment (and foreseeable future) it is simply not acceptable to route production traffic between two VoIP systems that have not undergone extensive interop testing between one another. I'm not just talking about vendor certifications, either. Almost every platform offers enough deviation in software revision and configuration alone to warrant testing between specific networks (and specific known configs at that). Even with adequate testing, how will my media be routed? Between endpoints over the internet? Over some other layer 3 peering fabric? What about potential issues in that transport? There needs to be something like an Acme to scrub the traffic between two incompatible (or suspect) peers. Simply using a SIP proxy as demonstrated in the RFCs (Alice, meet Bob) is a recipe for disaster (Bandwidth.com). On Wed, Jul 14, 2010 at 2:35 PM, Alex Balashov <abalashov at evaristesys.com> wrote:
One argument that's possible to make is that standardisation and commoditisation of all the relevant features has to some extent already--and will in the future further--reduce the need for this concern.
In other words, it's more and more common to find CPE and network elements that support any conceivable ITU-T codec, any reasonable ptime, all forms of DTMF relay, etc.
Obviously, we're not there yet. ?But I would wager that those capabilities are going to converge faster than it would take for any of the big operators to overhaul their networks with outboard transcoding gear or additional feature cards in every POP and so on.
-- Alex Balashov - Principal Evariste Systems LLC 1170 Peachtree Street 12th Floor, Suite 1200 Atlanta, GA 30309 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ _______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops
-- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com