[asterisk-users] Vitelity offline?
Vitelity seems to be offline to both IP and voice traffic. Is there any place to find out what their status is? Roger Marquis -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ITSP's no longer supporting IAX?
Steve Totaro wrote: I understand you are a developer and you want IAX2 to be great. That is your job, but the fact is that it is not and has caused audio and security problems for YEARS in EVERY release. It should bug you and everyone at Digium that waves the IAX2 flag. Can you elaborate on these audio and security problems Steve? Looking at the two protocol specs I cannot see a basis for your claim. IAX doesn't embed the local IP address in the packet data but that's surely no substantive security. It does separate data and signaling at the application-level, but again, that's no basis for such a claim. Protocols must be looked at separately from their implementations. From the various responses it appears that Asterisk 1.4's implementation of IAX has flaws. These do not necessarily reflect on the protocol. OTOH, there are a lot of engineers with SIP skill and experience who, naturally, are concerned with their investment in time, education, and experience. While this may or may not apply to Sonicwall engineering, it's also true that any streaming protocol will be better handled by devices that process packets in ASICs (high-end firewalls) rather than CPUs (PCs and low-end firewalls). FWIW (2 data points) I get uniformly better service from our IAX trunk provider than our SIP trunk provider. No idea whether that's protocol, implementation (1.4 on my side), or provider-related though I suspect the later. Roger Marquis ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ITSP's no longer supporting IAX?
Jon Pounder wrote: This sounds like a bunch of gobbledegook spewed out by those very high end firewall vendors. Call it what you want but anything that processes packets in any way and makes a decision on what to do is by definition a CPU. You won't find much support for that opinion in network engineering circles. The processing advantage of ASICs is easily measured and widely documented. ASICs are particularly critical to latency-sensitive protocols and those using small packet sizes with correspondingly high packet counts. According to Praveen Kumar (Founder/CEO of Packet Island) the ASIC differential is even more noticeable with interactive streaming video than streaming audio. Roger Marquis ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 client for 'eee pc 1000'
Rob Hillis wrote: The solution for the problem of an IAX client is a SIP client. That's not a particularly good solution if you have a NAT between your client and Asterisk. IAX is still *much* easier to get working through a firewall. It's working fine here (Twinkle/Ubuntu over NAT/Netscreen). Didn't have to change any settings on the firewall either. Roger Marquis ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QOS for outgoing SIP calls
Chris Mason wrote: QOS can only be on outgoing, you can't set the priority of a packet after you receive it. The only other solution would be the cooperation of the ISP to provide QOS upstream of you. Good luck. QOS is probably not the most precise term as it's normally associated with RSVP, MPLS, packet headers, etc. But you can, in Netscreens at least, define a Guaranteed Bandwidth. We do this for SIP/IAX IPs, in both outgoing and incoming policies, and it works both ways. Audio quality is good and there are no chan_sip.c: Peer is now (UNREACHABLE|Lagged) messages even during long DVD or Bitorrent xfers. The reason it works outbound is a no-brainer, but inbound bandwidth is also effectively guaranteed. Sure there's no way to control external devices that ignore ICMP source-quench or break TCP congestion control but those flows are typically limited to nefarious sources which would not be responsive to other types of QOS anyhow (BGP being one potential exception). Roger Marquis ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
