On this subject of sip providers, how do you guys feel about Broadvox as a provider? Any good things about them, bad? I've been testing with them for a couple of weeks and so far the quality has been great for me.
Thanks for anyone's input, Tom -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lane Sullivan Sent: Sunday, June 08, 2008 2:45 AM To: [EMAIL PROTECTED]; 'Commercial and Business-Oriented Asterisk Discussion' Subject: Re: [asterisk-biz] SIP providers Trixter, Thanks, I have all email leaving my mail server append a footer. I didn't think about it appending to messages on this list. I have eliminated it for this list. Sorry again. :) On the other topic. Can I just say implementing VOIP can be easy but doing it RIGHT requires some insight and planning. I am planning my first implementation so any advise would be great. Sometimes learning what people would do differently is easier than reinventing the wheel! Thanks, Lane -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Trixter aka Bret McDanel Sent: Saturday, June 07, 2008 11:27 PM To: Commercial and Business-Oriented Asterisk Discussion Subject: Re: [asterisk-biz] SIP providers On Sat, 2008-06-07 at 22:56 -0600, Lane Sullivan wrote: > I am just wondering what provider everyone uses for their SIP trunking > and their thoughts on who they use. > > First, sip trunking has some standards in the works but almost no one implements them. A side note before someone else jumps in, sip is just signalling, rtp carries the media, and from its initial designs was allowed to have trunking capabilities, but most rtp stacks disregard this capability in favour of a much simplier method. In general its one RTP port per RTP stream. This *can* be more efficient, depending on the model used (ie if each rtp stream has its own thread its generally better especially on a multi-cpu/core system, than if all of them go into one port since only one cpu/core can service that one port). Section 5.2 of RFC1889 (RTP) discusses some of the drawbacks of trying to do trunking with that variant of RTP (the standards I mentioned are elsewhere and use a slightly different method for delivery, basically its an CRTP/RTP tunnel) As a result not many do trunking with sip. Unless you meant trunking in a different sense, and not the aggregation of multiple RTP streams into one "connection" (yeah yeah udp doesnt do connections). Lastly, you may want to have an alternate signature if possible for lists. Yours is quite long, and generally anything over just a few lines is considered excessive. If you think about it, your signature is almost 850 bytes, that gets sent to thousands of people turning what seems to be a simple email into a couple megs of traffic, for each email. And the fact that its several times the size of your actual content makes it stand out a little more I think. > -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast +44 28 9099 6461 US +1 516 687 5200 http://www.trxtel.com the phone company that pays you! _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
