An old but interesting article from the Sydney Morning Herald -
German police are unable to decipher the encryption used in the internet telephone software Skype to monitor calls by suspected criminals and terrorists.
"The encryption with Skype telephone software ... creates grave difficulties for us," Joerg Ziercke, president of Germany's Federal Police Office (BKA) told reporters, "We can't decipher it. That's why we're talking about source telecommunication surveillance -- that is, getting to the source before encryption or after it's been decrypted."
Experts say SVoIP software are difficult to intercept because they work by breaking up voice data into small packets and switching them along thousands of router paths instead of a constant circuit between two parties, as with a traditional call.
Source Article here.
Friday, May 16, 2008
Internet telephone encryption stumps police
Saturday, December 1, 2007
Expert scares world with VoIP hacking proof
From Techworld -
An expert, Peter Cox, co-founder and ex-CTO of firewall vendor BorderWare, has released a proof-of-concept program to show how easy it would be for criminals to eavesdrop on VoIP-based phone calls.
Called SIPtap, the software is able to monitor multiple VoIP call streams, listening in and recording them for remote inspection as .wav files. All that is needed would be to infect a single PC inside the network with a Trojan incorporating these functions, although the hack would work at ISP level as well. The program can index 'IP-tapped' calls by caller - using SIP identity information - and by recipient, and even by date.
SIPtap demonstrates that the worst-case nightmares of VoIP vulnerability are now well within the capabilities of organized crime, which could use such a program to steal confidential data from companies, governments and even the police.
Labels: VoIP
Thursday, August 30, 2007
Bugging VoIP SIP Phones
From Sunnet Beskerming -
Recently findings suggests that it is a relatively simple matter to remotely eavesdrop on a broad range of SIP-enabled devices. For readers who aren't aware of what SIP-enabled devices are, SIP (Session Initiation Protocol) is a protocol that is used by a lot of VoIP software and associated telephone handsets to establish, modify, and control a VoIP connection between two parties.
The research that was published indicates that, for at least one vendor, it is possible to automatically call a SIP device from that vendor and have it silently accept the call, even if it is still on the hook - instantly turning it into a classic bugged phone. Whereas historic telephony bugs needed physical targeting of the line running to a property or place of business, the presence of VoIP in the equation allows bugging from anywhere in the world with equal ability. Now anyone can do from their armchair what only spies and law enforcement used to be able to do from inside the telephone switch / pit / distribution board, though it's still illegal to do so.
As well as bugging the phone, the action effectively acts as a Denial of Service against the device (after all, it is already engaged in a call).
Having found the bug via fuzzing, the discovering researchers believe that there may be a number of vendors that have created their own SIP networking code, with equivalent bugs contained within.
While the vendor concerned is expected to release appropriate patches soon, the disclosure is likely to turn attention on other SIP device providers.
This may already be happening, with two separate exploits released publicly in the last couple of days targeting Cisco SIP handsets, with the result of a Denial of Service condition against the phones. VoIP client software from eCentrex has also been targeted with public exploit code, except this time it allows for control over vulnerable devices as a result of a remote buffer overflow condition.
Concerned users and administrators who have SIP enabled software or hardware should be aware of their potential limitations and have appropriate mitigation strategies in place, especially if they are used in sensitive areas (military use, national secrets, trade secrets, etc).
Labels: VoIP
Thursday, June 14, 2007
Is wireless [Wi-Fi-based] VoIP worth the bother?
From Network World -
Per the article, the answer for enterprise deployment is generally negative. However, recent trends provide hope.
Morever, successful deployments though still rare at this time, such as Osaka Gas in Japan, provide compelling proof of VoIP's viability. The utility used Meru Networks’ USD10M WLAN infrastructure to support 6,000 mobile phones that were equipped with cellular and Wi-Fi network interfaces.
The reluctance to embrace large-scale wireless VoIP isn’t suprising. Enterprisewide wireline VoIP deployments have only fairly recently found traction, and many of these have been angst-ridden. To be fair, often the angst is created by specific issues or problems at a given enterprise site.
But using a wireless connection in place of a wire adds lots of complexities, solutions to which are only slowly maturing. Access points have to be pervasively distributed to support voice traffic, while radio interference can easily affect voice quality or call sessions. Wireless eavesdropping on unsecured VoIP sessions is another worry for enterprise managers.
And it’s difficult to pinpoint savings since corporate IT doesn't have a good handle on actual spending since some of the equipment have other functions. Thus it is often just expensed, making it difficult to make a case for savings and therefore to make a case for investing in VoIP over WLAN.
Over the course of three months WLAN switches and access points from Aruba Wireless Networks, Chantry Networks (now Siemens), Cisco and Colubris Networks in terms of audio quality QoS enforcement, roaming capabilities, and system features were tested. Among the findings:
* With QoS enforcement turned on, and with only voice traffic on the net, calls nearly matched toll-quality audio.
* With even a small amount of data traffic, dropped calls became common and audio quality was poor, even with QoS still enabled.
* Roaming from one access point to another either failed or took so long, from 0.5 to 10 seconds, that calls dropped.
Those findings reflect some of the experience at Dartmouth College, which deployed a limited VoIP system on its pervasive Aruba-based campus WLAN four years ago. Initially, some college staff used the wearable mobile VoIP phone from Vocera. There were some problems with roaming, according to David Bucciero, Dartmouth director of technical services, who despite these teething pains is one who says wireless VoIP is worth the hassle.
More recently, the college has added just under 100 Cisco 7920 wireless VoIP handsets which “were flawless,” though latency was an issue early in the deployment. Reducing those delays has been an ongoing tuning process.
Things have changed in two years, including the advent of the 802.11e QoS standard, augmented by continued proprietary QoS tweaks, and faster handoffs between access points.
But the real change has been the growing interest in, and products for, shifting call sessions automatically between cellular and Wi-Fi. At the enterprise level, this convergence entails an IP PBX, usually a SIP server, the WLAN infrastructure, new specialized servers from start-ups like Divitas and established players like Siemens, and accompanying client code running on so-called dual-mode handsets.
Dartmouth is doing exactly this, running a pilot test with the Nokia E61i which uses SIP to talk to the Cisco CallManager IP PBX.
“Cellular and Wi-Fi convergence is the real pull for VoIP over wireless LANS,” says Farpoint’s Mathias. “Once that [convergence] happens, then we can converge dialing directories, voice mail, other services, and have one phone that works everywhere.”
Labels: VoIP
