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Regarding Matter of Quality
by CCurry＠telovations.com Jan. 20, 2011

Jan. 20, 2011

We use Palladion from IPTEGO. We looked at the entire market vendors for monitoring tools and this was head and shoulders above the rest. Specifically addressing your requirements, all calls and speech quality measurements and also the actual RTP if you want, is recorded in the database going back days or months if you require. So you can go back and find dropped calls. It will specifically detect the dropped call if signaling is not seen for a specified number of seconds. Alternatively you could look for a dropout of speech quality In addition to your other requirements, it obviously does MOS, packet loss and jitter etc. It has a nice message flow diagram, to trace a call from different geographic locations in your network. This is accessed from a simple drill down from the summary information. Also it can scale to very high volumes. It uses standard Linux hardware, so you can drop another quad/HEX core blade into your box and you are good to go unfortunately, it doesn't make the coffee. But it's cheaper than running these measurements on a network device such as an SBC etc. [cid:image001.png at 01CBB8A3.414DF840]

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Matter of quality
by sil＠infiltrated.net Jan. 20, 2011

Jan. 20, 2011

For those on this list who'd be willing to share particularly from say an ITSP (Carrier) side, off-list, on-list, etc., what are the percentages for quality, system issues some of you are using for baselines/metrics. For example, I have one or two (alright maybe 10-20 at max) common 's:whiner:clients:g' who have caught management's ear. "My call to a cellular dropped after 120 minutes!@" From my POV, I have two maybe three common clients who are complaining... "Dropped call, call sounded garbled, etc., etc., etc" From my perspective, even if I had 1,000 of these calls in a day, it's less than .5% of the amount of calls I process. Nevertheless, we as engineers want to perform due diligence versus blurting the ever popular "told you so." This means tracking down all other carriers' issues (Comcast filtering anyone), any potential outages, and so on (you all know the routine). So long story (rambling) short, what are some of the gauges you guys (big boys not onesies twosies) use to measure quality metrics (amount of dropped calls versus successful calls, ARS ratios, MOS, etc). I use a combo of Acmes, NCites and soon to be Transnexus in the mix. Also, what pro-grade measurement tools are recommended, less I waste an hour @ CAIDA or similar. Requirements of tools in mind: MOS would be nice, tracing calls would be nice, storing SIP sessions would be cool. High volume traffic parsing (think 100s of gigs here) Making coffee, even cooler! (Beggars can't be choosy). For the vendors on the list selling tools, please don't solicit, I'd rather hear what peers have to say rather than spend time on the phone away from the beauty of figuring out why someone else's configuration/setup doesn't work. =+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ J. Oquendo SGFA, SGFE, C|EH, CNDA, CHFI, OSCP, CPT "It takes 20 years to build a reputation and five minutes to ruin it. If you think about that, you'll do things differently." - Warren Buffett 227C 5D35 7DCB 0893 95AA 4771 1DCE 1FD1 5CCD 6B5E http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x5CCD6B5E

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RTP issues to Dash Atlanta SBC
by chris.ledford＠cnxntech.com Jan. 12, 2011

Jan. 12, 2011

Sorry all I meant to say RTP drops to the Atlanta SBC. I had a runaway tcpdump running when I wrote it :) . My Bad. V/r Chris Ledford CCNA VOICE/CCSP Comptia A+/Net+/Linux+/Sec+ EWC/CTTC(sw) USN T3 Engineer

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VoiceOps Digest, Vol 19, Issue 9
by chris.ledford＠cnxntech.com Jan. 12, 2011

Jan. 12, 2011

We are currently experiencing RTP to Dash's Atlanta SBC. Is anyone else seeing this? V/r Chris Ledford CCNA VOICE/CCSP Comptia A+/Net+/Linux+/Sec+ T3 Engineer http://navy.togetherweserved.com/profile/13552

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Centralized management for distributed Asterisk servers
by robert.dawson＠mindshift.com Jan. 11, 2011

Jan. 11, 2011

I have a customer that wants to maintain a standalone asterisk server at each of roughly 20 sites with a centralized configuration engine. Something similar to what Fonality (I think?) used to offer. They simply will not go with a centralized/hosted solution. Anyone know of anything similar to this? Thanks, Rob

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What is the State of the Art in VoIP Network Design?
by lindsey＠e-c-group.com Jan. 11, 2011

Jan. 11, 2011

It's best for the Science and Art to have a public discussion about VoIP network designs. What was once an incidental installation task is often recognized as a key activity with serious cost implications. But still, many networks are never designed -- they just happen. IP Networks do not -- by default -- work well for VoIP. We the industry should document the current state of the art for the design of VoIP Networks. The need is greatest in carriers and large enterprises networks, but the design of networks for VoIP at all scales is interesting. Please help me document the standard industry practices in VoIP Network Design, both of new networks and also the expansion and evolution of existing networks. Your responses will be analyzed and compiled into a publication that will be submitted for conference publication. I hope to hear from vendors such as Acme Packet, Audiocodes, BroadSoft, Cisco, Digium, Genband, Metaswitch, Sonus, Taqua, and others. But I especially want feedback from individual designers & integrators. Everything you provide will be published online and credited. Responses received by February 15, 2011 will be used for the paper submission. My employer, ECG, Inc. is a for-profit entity, but this project is NOT a commercial enterprise. We wish to foster and sponsor innovation in VoIP networks in an open, free academic dialog. Mark R Lindsey, ECG Inc. mark at ecg.co | +1-229-316-0013 | http://ecg.co/lindsey ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ VoIP Network Design Survey. Please respond to lindsey at e-c-group.com A. If you would be willing to submit reference network designs for publication, please send them in PDF format. [VNDS1-A] B. Please provide answers to the following questions -- as much as possible. Even if you only have time to answer one question. [VNDS1-B] Q-1. What application-layer (VoIP) network devices do you regularly design with? What are special network requirements of each device? [VNDS1-B-1] Q-2. Where do you use public IP addresses, and where do you use private IP addresses? In each case: why do you use that type of address? [VNDS1-B-2] Q-3. How do you handle customer-premise access devices behind NAT? [VNDS1-B-3] Q-4. How do you handle peering ("SIP trunks") between carriers? (e.g., SBC? Separate IP assignment per endpoint? Separate Port assign nment? Authentication?) [VNDS1-B-4] Q-5. How do you guarantee network capacity for VoIP Media traffic? ...for VoIP signaling traffic? [VNDS1-B-5] Q-6. Are you designing to support SIP over TCP or SCTP? If so, why? If not, do you address fragmentation of large SIP? [VNDS1-B-6] Q-7. How do you describe your approach to network security boundaries? How do you enforce these boundaries? (Common approaches: firewalls, SBCs, VRFs/routing domains, server packet filters, private IPs and routing policies, etc.) [VNDS1-B-7] Q-8. What are the key security or operational threats that you design to protect against? [VNDS1-B-8] Q-9. What are the key network design principles for VoIP Networks that you follow? [VNDS1-B-9] Q-10. What are the key mistakes you try to avoid in VoIP Network design? [VNDS1-B-10] Q-11. What future trends in network design and requirements do you anticipate? [VNDS1-B-11] Q-12. What are your tightest constraints in your network designs? (e.g., Design time? Testing/Stability/Change Control? Rack space? Network link capacity? Money? CPU Capacity? CAM space?) [VNDS1-B-12]

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Adtran 924 PRI
by lists＠iamchriswallace.com Jan. 11, 2011

Jan. 11, 2011

We typically don't have issues with these lines. We just make sure to force G.711 and enable modem pass-through. ---Chris On Jan 10, 2011, at 7:58 PM, Dane Jasper wrote: > What about support of credit card processing machines? We've got Adtran TA5000 units deployed serving POTS which seem to be incompatible with many units. > > -Dane > > > Chris Wallace wrote: >> Great box and you can't beat the price, handles both PRI and CAS T1. Also supports T.38 on the analog ports and works quite well now that the software bugs are worked out. We have well over 100 of these units deployed and have no complaints. >> >> ---Chris >> >> On Jan 2, 2011, at 9:35 PM, Eric Hiller wrote: >> >>> Does phone equipment see the Adtran 924 PRI as a true PRI? Ie. can this be >>> used in place of a true PRI? Are there any hangups I should know about? I >>> was just going to use this to provide PRI to a customer over SIP rather >>> than have to run a PRI as well. >>> >>> Any advice with the 924? >>> -Eric >>> >>> _______________________________________________ >>> VoiceOps mailing list >>> VoiceOps at voiceops.org >>> https://puck.nether.net/mailman/listinfo/voiceops >> >> >> _______________________________________________ >> VoiceOps mailing list >> VoiceOps at voiceops.org >> https://puck.nether.net/mailman/listinfo/voiceops >> > > -- > Dane Jasper > CEO & Co-Founder > Sonic.net > Sonic Telecom > dane at corp.sonic.net > 707-522-1000 x2205 >

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Growing attack pains
by sil＠infiltrated.net Jan. 10, 2011

Jan. 10, 2011

I'm in the market for something to place in front of an SBC (modules would be nice, e.g., Asterisk module, Avaya module, etc.) The device will need to do the following: Block on N ... Block N amount bad attempts indefinitely and alert Block on Prefix ... If PREFIX is anywhere in SIPURI/ANI/CID, block (country specific would be nice) We are having a hard time keeping up with the attack vectors here. We recently saw a compromise from Egypt where the password was 15 characters mixed numbers, letters and symbols. So obviously longer passwords aren't even an issue anymore. -- =+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ J. Oquendo SGFA, SGFE, C|EH, CNDA, CHFI, OSCP, CPT "It takes 20 years to build a reputation and five minutes to ruin it. If you think about that, you'll do things differently." - Warren Buffett 227C 5D35 7DCB 0893 95AA 4771 1DCE 1FD1 5CCD 6B5E http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x5CCD6B5E

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Adtran 924 PRI
by clec＠cygnustel.com Jan. 10, 2011

Jan. 10, 2011

Does phone equipment see the Adtran 924 PRI as a true PRI? Ie. can this be used in place of a true PRI? Are there any hangups I should know about? I was just going to use this to provide PRI to a customer over SIP rather than have to run a PRI as well. Any advice with the 924? -Eric

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Combination bug EdgeMarc/Polycom?
by owen＠impulse.net Jan. 9, 2011

Jan. 9, 2011

Hello Voice-Ops, I have a frustrating situation where a random 1% of PolyCom phones (Mix, but all running 3.1.1) behind EdgeMarcs (VRRP 9.1.9) configuration are losing registrations overnight after a failed boot server check-in. (FTP) Phone will be URL call disabled until reboot or 12 hour+ wait. Inbound calls still work, outbound calls fail. Registration from a packet level is normal. Scenario is complex, but repeatable at multiple client sites. (so not firewall/lan) wireshark, tech support, release codes, ftp logs, and boot logs are all unhelpful/empty, and the firmware code (s) we're running can't be changed without proof, which at this point can only be generated by a fun weekend spent rebooting a cluster of phones over and over again, which for an annoying bug seems excessive. Bit of a rim-rock. Anyone seen anything like this? Cheers, Owen R. ____ Senior Network Engineer Impulse Advanced Communications 805-884-6332 owen at impulse.net

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