Re: [VoiceOps] Homer vs VoIPMonitor

I believe you are correct that Homer only does SIP, where VoIPMonitor does both. This bring me back to the question of Polycom Phones with VQMON that use thee standard *RTCP-XR (IETF RFC 3611) format and can publish metrics using the SIP PUBLISH method. These are RTP stats for the most part

Re: [VoiceOps] Homer vs VoIPMonitor

On 20/11/15 14:22, Colton Conor wrote: > I believe you are correct that Homer only does SIP, where VoIPMonitor > does both. The latest version of homer capture agent can track rtcp: https://github.com/sipcapture/captagent Cheers, Daniel -- Daniel-Constantin Mierla

Re: [VoiceOps] Homer vs VoIPMonitor

+1. On all points. Thanks, Shripal > On Nov 19, 2015, at 11:03 PM, jay binks wrote: > > Ive used Voipmonitor for a few years now, it started out OK, but now.. its > AWESOME. > > parts are open source ( mainly the sniffer / analyser ). > the commercial part is the web

Re: [VoiceOps] Homer vs VoIPMonitor

I have had experience with Empirix, Palladian and VoIPMonitor. Both Empirix and Palladian are EXPENSIVE and very hard to Engineer correctly to capture all the messaging. An under-engineered system will be VERY annoying and a HUGE waste of money. VoIPMonitor has been extremely valuable, very

Re: [VoiceOps] Homer vs VoIPMonitor

Jumping on the VoIPmonitor bandwagon here; they've been more responsive than the average vendor to new feature requests. A memorable interaction went something like this: Voxox: "We'd like voipmon to do X." (e.g. process RTCP-XR and SIP PUBLISH reports from Polycom, or analyze a proprietary

Re: [VoiceOps] Preventing random SIP connections to handsets

On 11/20/2015 12:14 PM, Carlos Alvarez wrote: > We're starting to see customers who get random arbitrary ringing caused by > a random connection attempt from the internet. Most of our customers have > Cisco routers with full-cone NAT, so it's easy to do that. We don't > reinvite handsets, we

[VoiceOps] Preventing random SIP connections to handsets

We're starting to see customers who get random arbitrary ringing caused by a random connection attempt from the internet. Most of our customers have Cisco routers with full-cone NAT, so it's easy to do that. We don't reinvite handsets, we proxy the media, so we've considered using restricted NAT

Re: [VoiceOps] Preventing random SIP connections to handsets

We have a Calix ONT in our lab that is ‘on the internet’ for its voice VLAN. It gets rogue INVITES and rings constantly (every 5-10 seconds). Makes for a nice honeypot, source IPs go right into the ACL on the firewall — Matthew Crocker President - Crocker Communications, Inc. Managing

Re: [VoiceOps] Preventing random SIP connections to handsets

These routers use a range of high ports, nothing is on 5060. It seems that they are scanning, and when they get something, repeatedly attack it. Because our enterprise customers are all on a couple of subnets, they may have keyed into "this range is full of SIP devices" and keep hitting them.

Re: [VoiceOps] Preventing random SIP connections to handsets

I was getting ghost ringing into my Polycom because my router sensibly remaps phone:5060 to WAN_IP:5060. My solution was to switch to SIP TCP. On 11/20/2015 03:14 PM, Carlos Alvarez wrote: We're starting to see customers who get random arbitrary ringing caused by a random connection attempt

Re: [VoiceOps] Preventing random SIP connections to handsets

On 11/20/2015 03:23 PM, Carlos Alvarez wrote: That's the default for all the handsets, I believe. There are various options such as "accept only from proxy" or "only from registrar," but like I said it varies so it could be more challenging to employ that. Also in our limited testing it seems

Re: [VoiceOps] Preventing random SIP connections to handsets

> On Nov 20, 2015, at 3:27 PM, Alex Balashov wrote: > > On 11/20/2015 03:23 PM, Carlos Alvarez wrote: > >> That's the default for all the handsets, I believe. There are various >> options such as "accept only from proxy" or "only from registrar," but >> like I said

[VoiceOps] Charter

Are there any Charter contacts on the list? We're seeing 100% packet loss in ATL. Thanks David -- David Wessell President, Ringfree Communications, Inc p: 828-575-0030 x101 a: 475 S Church St Hendersonville, NC 28792 s: www.ringfree.com e: da...@ringfree.com

Re: [VoiceOps] Preventing random SIP connections to handsets

That said, it is baffling to me that phones accept INVITEs unchallenged from anywhere but $REGISTRAR_IP/$OUTBOUND_PROXY_IP. -- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA 30346 United States Tel: +1-800-250-5920 (toll-free) /

Re: [VoiceOps] Preventing random SIP connections to handsets

That's the default for all the handsets, I believe. There are various options such as "accept only from proxy" or "only from registrar," but like I said it varies so it could be more challenging to employ that. Also in our limited testing it seems like it may not have had the intended effect.

Re: [VoiceOps] Homer vs VoIPMonitor

Is anyone clustering VoIPMonitor? Geographically diverse networks with probes reporting up to a single front end? --- Brian J Murray -Original Message- From: VoiceOps [mailto:voiceops-boun...@voiceops.org] On Behalf Of Calvin Ellison Sent: Friday, November 20,

Re: [VoiceOps] Preventing random SIP connections to handsets

We've never seen evidence of issues other than just making phones ring. I assume it's some script kiddies trying to find an open SIP proxy. The routers in use are owned by our ISP partner and managed by them. Typical mid-grade routers like a 1900 series. I'm not aware of an ability to filter

Re: [VoiceOps] Homer vs VoIPMonitor

Yup! Remote probes solved a couple challenges: Not only enabling geographically diverse networks, but also removing the need for port mirroring/SPAN within a datacenter. Regards, Calvin Ellison Voice Services Engineer calvin.elli...@voxox.com +1 (213) 285-0555

Re: [VoiceOps] Preventing random SIP connections to handsets

On Fri, Nov 20, 2015 at 12:21 PM, Alex Balashov wrote: > That said, it is baffling to me that phones accept INVITEs unchallenged from > anywhere but $REGISTRAR_IP/$OUTBOUND_PROXY_IP. ^This. Challenge to INVITE should mitigate scanners, and challenge to BYE should

Re: [VoiceOps] Preventing random SIP connections to handsets

On 11/20/2015 04:09 PM, Calvin Ellison wrote: challenge to BYE should mitigate that particular targeted attack. Spoofed sequential (in-dialog) requests strike me as less of a concern than initial requests, since, in order for the BYE to match an existing dialog in the phone's UAS, the

Re: [VoiceOps] Homer vs VoIPMonitor

We have multiple probes on the east coast and west coast, reporting to a front end server. Works great! Sean Salvadalena | Network Engineer -Original Message- From: VoiceOps [mailto:voiceops-boun...@voiceops.org] On Behalf Of Brian Murray Sent: Friday, November 20, 2015 1:01 PM To:

Re: [VoiceOps] Homer vs VoIPMonitor

Anyone have a sales contact at voipmonitor they can share or should I just go through their webpage? Feel free to reply on or off-list. Thanks, Pete On Nov 20, 2015, at 16:33, Gavin Henry wrote: On 20 Nov 2015 21:15, "Calvin Ellison" wrote:

Re: [VoiceOps] Homer vs VoIPMonitor

Just email i...@voipmonitor.org. Martin will respond quickly. Geoff Mina President & CIO Connect First, Inc. 720.335.5924 Website | LinkedIn | Email On Nov 20,