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VoiceOps Digest, Vol 24, Issue 23
by tallen＠acmepacket.com June 23, 2011

June 23, 2011

Have you looked into the line of products from Integrated Research? I think that their IP Telephony Assessor might meet your needs http://www.prognosis.com/Products_and_Solutions/IP_telephony_monitoring/Pro…. Cheers, Toby

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Active Testing Probes -- Tektronix?
by lindsey＠e-c-group.com June 23, 2011

June 23, 2011

I'm still on the hunt for a robust active-probing tool for VoIP providers. Does anyone have experience with the Tektronix PowerProbe 500 or 6000 probes? Are they still sold? Tektronix' sales literature is dated 2008. http://www2.tek.com/cmswpt/tidetails.lotr?ct=TI&cs=afs&ci=14603&lc=EN I want devices that can: (a) Register (b) Place and receive calls through an SBC. (c) Function properly behind NAT, with SBC help. (d) Place and receive POTS (two wire analog) calls, and report statistics through the network interface (e) Run calls on a schedule (which implies that remote probes connect in to get instructions.) (f) Report statistics on every call: (1) For VoIP calls, RTCP-XR -like-stats are good. (2) For POTS calls, eMOS / PESQ is nice, plus more precise measurements, such as signal-to-noise ratio, echo, audio round trip time, frequency-response spectrum, or other audio defects. Ideally, do all this with a Linux command-line interface for me and a pretty web interface for anybody who asks. mark at ecg.co | +1-229-316-0013 | http://ecg.co/lindsey

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Engineering SIP Timers: Interview with Sam Rausch, Comcast
by lindsey＠e-c-group.com June 21, 2011

June 21, 2011

If you didn't make it to SIPNOC 2011, you missed a great talk by Sam Rausch of Comcast. He recorded an interview with me, and told me about Comcast's SIP Timer Engineering project. Many VoIP service providers have just given up on failover between SBCs at different locations with different Untrusted IP addresses. Sam tackles this head on and reports good progress. The interview is now edited and online as a Podcast. You can download the 15 minute interview here: http://www.200ok.info/ mark at ecg.co | +1-229-316-0013 | http://ecg.co/lindsey

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Porting Number - CLEC
by clec＠cygnustel.com June 20, 2011

June 20, 2011

In order to port a number, what trunking requirements do we have? Is it that we have to have trunking to the same tandem as the npa-nxx holder? IE. If I wanted to port 100-100-1000 And the 100-100-1 thousands block is held by company A, and then 100-100 npanxx/ten k block is held by Company B and is homed on Tandem C, then I just need to have trunking into Tandem C and have an ICA with whomever controls Tandem C, which could be company A,B or some third company C correct? The same would apply in order to get our own thousands block as well correct? Thanks, -Eric Hiller Cygnus

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SBC's that drop traffic based on domain
by CCurry＠telovations.com June 17, 2011

June 17, 2011

In an effort to mitigate DDOS attack's I am trying to deny all traffic based on the request-uri host domain. The reason being from what I see is "most" attacks are sent to the SBC's IP address and does use the domain name. When the proper domain is supplied I would like to allow that packet. All other I will not respond to period. Example of hacker Requet URI Ex. INVITE sip100:199.44.55.22 SIP/2.0 Legit Request URI Ex. INVITE sip:7724558787 at voip.myvoice.net SIP/2.0 I have tried to create an HMR on ACME with little success. I can get the registers to not respond yet only if sip:199.44.55.22 is use. If the attacker uses sip:100 at 199.44.55.22 the SBC still will respond with a 403. Besides that All invites are always responded to regardless even though the HMR(Header Manipulation) should be using Invite and registration meathods. I have tried to get ACME to come up with a solution yet have been unsuccessful. They will not even take my request for a feature enhancement. Has anyone had any successful experience at implementing this on any other SBC platform? I know there are many ways to protect yourself from DDOS attacks yet to me this is a simple first line of defense. [cid:image001.png at 01CC2C46.97470A90]

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Equipment Sale.
by alejandro＠amerimex.com June 15, 2011

June 15, 2011

All, We are AmeriMex Communications Corp. CLEC based in Roswell, GA, we have three collos in AT&T's GA area (Roswell, Cumming and Gainsville), we have Adtran TA3000 and TA5000 installed and working with capacity of 500 POTS in each one of those COs. Roswell equipment also supports T1s. We are selling all equipment if you are interested please contact us. alejandro at amerimex.com Thanks, <outbind://25/AmeriMexSignature_files/image001.jpg> Alejandro Caipa IT Director 1007 Mansell Road Suite A Roswell GA 30076 Direct Dial: 678.832.6220 alejandro at amerimex.com www.amerimex.com

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ACME behaviour with Bad NAT Firewalls
by ujjval＠simplesignal.com June 10, 2011

June 10, 2011

Hi All: We have seen many PBX NAT?ed behind a Firewall that does not do the Sip ALG correctly. Most cases putting the Private IP in the Contact header and the ACME responds back to the Private IP. Example call inbound to the PBX, the PBX sends a 200 OK (as call is answered) with a Private IP in the contact header. ACME sends the ACK back to the Private IP blackholing it. I have seen SBC?s that do adjust to the Layer 3 IP:port if they notice Private IPs in the SIP signaling. Is there a setting on ACME to do that? Thx, UK

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Polycom ghost ringing
by peeip989＠gmail.com June 7, 2011

June 7, 2011

Back in January there were a couple of posts related to "ghost ringing" on Polycom phones. At the time the workaround seemed to be to downgrade to v3.3.0 (we're on 3.3.1). Looking at the release notes on 3.3.1 it looks like the .0 release was pretty buggy and therefore we'd rather not downgrade to it. Polycom has not acknowledged (to us) that they even know about the problem. Does anyone know if any headway has been made in actually fixing the issue?

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SS7 analysis
by frnkblk＠iname.com June 7, 2011

June 7, 2011

Every once in a while we have an incoming calling issue (TDM side of the house) where we need to talk to our tandem to find out which trunk of theirs the call came in on. Most of the time we have to do that while the call is in progress, or wait till it comes in again so they have their SS7 analyzer ready. A person in our company heard from another telco who uses another tandem that it's possible for us not to involve our tandem, but discover/learn at least some part of the call path using SS7, that is, be able to see beyond our tandem. Our tandem says that's not possible, but our person says the other telco is sure about this. Anyone know either which way? Frank

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SIP calls that aren't torn down
by frnkblk＠iname.com June 7, 2011

June 7, 2011

Over the last few months our softswitch has accumulated 10 "stuck" calls where there's no media traffic. From what the softswitch vendor can tell or guess, it just didn't receive a BYE to tear down the call. We know that most SIP traffic on 5060 is UDP, and UDP is connectionless. How are other vendors and systems managing such scenarios? I suggested a "no media" test where after x hours of no media, to tear down the call and log the info. Thoughts? Frank

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