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August 27 - Issue 50
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MCS Forum is an independent forum on
Microsoft Communications Server & UC-Unified Communications.
Microsoft is a trademark of the Microsoft Corporation.
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Greetings!
Welcome to the MCS Forum.
A new look for a new era in unified communications. In you would like to see your products highlighted, reviewed and presented here or in other publications such as:
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AudioCodes SBC - Designing Secure SIP-OCS Networks Using SBCs
In this next part of a multi-part series, we will examine
one of the fundamental tasks required of a SBC - Call Admission Control. Data Firewalls are not VoIP aware, and so are unable to selectively admit VoIP
packets. The SBC on the other hand is very voice aware. The SBC examines
incoming signaling and media packets and admits or blocks them based on
provisioned policies. Typical policies used in an Enterprise SBC
are:
- White List Policies: For
example, allow signaling packets from the Service Provider Call Server and the
IP-PBX at Headquarters only.
- Black List Policies: For
example, block all signaling packets from a set of Telemarketers domain names,
or from certain countries.
- Resource Management Policies: For
example, if there is a 100 session capacity, ensure that traffic from neither
the Service Provider or Headquarters individually consumes more than 70% of
those 100 sessions.
- Packet Screening Policies: For
example: block all packets over a certain size, allow only packets of specific
protocols, limit packet rates overall and from specific senders, etc
In addition to screening signaling packets, the SBC also
screens the media packets. In this role, the SBC must admit media packets
only for calls that have validly been established on the signaling plane -
having been through the policy screening outlined previously.
Media traffic is carried in RTP (Realtime Transport
Protocol) packets, and each media session is established on a different
port. RTP ports are contained in UDP-User Datagram Protocol, not
RTP. The RTP packet is carried as a payload in the UDP Packet ("DATA"
field). The RTP packet has its own header, and the media is it's payload.
There are a range of commonly used ports for SIP such as 5060-5061, however,
any port can be assigned which can cause blocking of calls. The SBC by
default blocks all RTP ports. RTP ports are only opened when selected for
use in a current call and send to the far end in the SDP (Session Description
Protocol) media negotiation. And when the call is terminated, the
associated ports are closed again. Not only are the ports dynamically
opened and closed as the associated calls are set up and taken down, the
packets admitted through them are limited to those coming from the end point
identified as the other end of the call in the SDP negotiation. Ports for RTCP, the companion protocol to RTP providing voice
quality statistics are also opened and closed in sync with their corresponding
RTP ports.
Before proceeding, an explanation of RTP and RTCP is
relevant to understanding how to manage an IP voice network. That is, in
order to perform the functions above, QoS and other functions, the function and
purpose of each element in the RTP-RTCP needs to be understood.
NOTE: IF YOU CAN'T READ THE SLIDES CLICK HERE
NOTE: Click here for the complete details on the RTCP-XR-MRB.
RTCP-XR-Real Time Control eXtended Reports is one of the key tools in diagnosing and
troubleshooting VoIP networks. XR-eXtended Report technology can be integrated into IP Phones or PSTN
Gateways. XR packets (see graphic for
packet format) are sent periodically during the call to provide real time
feedback on call QoS-Quality of Service. However, VoIP/SIP network planners need to consider the amount of XR
traffic that also consumes and reports on voice traffic. That is, diagnostic XR traffic consumes
bandwidth to diagnose traffic. The RTCP
MRB-Metrics Report Block provides measurements (metrics) for monitoring quality
of VoIP calls and conversations. These measurements include packet loss and
discard metrics, delay metrics, analog metrics, and voice quality metrics. The
Metrics Report Block reports individually on packets lost (discarded) on the IP
channel as opposed to packets that have been received and then lost by the
receiving jitter buffer. MRB reports on the combined effect of losses and
discards which can be used to determine corrective actions on voice QoS.
Thanks to AudioCodes for their help in this presentation. |
Cisco SBC and MTP - Redundancy-Resiliency & Scalability
Click here for animated tutorial
From the "must-read" book VoIP Performance and Optimization from click here Cisco Press ISBN 1-58705-528-7, the authors Ahmed, Madani and Siddiqui present, "The Session Border Controller (SBC)or the border element (BE) can provide another level of address masking while performing other tasks such as call filtering, call normalization (CODEC interoperability and fast-start and slow-start call setup methods) and bandwidth management. SBC/BE simplifies interoperability by allowing only one conduit to be opened for access to the aggregation point."
"The Cisco Unified Communications Manager (CUCM) and all the endpoints, including IP phones and gateways have private IP addresses. The SBC/BE (Border Element) and the MTP-Media Termination Point have public addresses. MTP and SBC functionalities can be offered in one physical device. Also, there can be several MTPs and/or SBCs for redundancy-resiliency and scalability. All the media from the SP-Service Providers network are sent through the MTP. There is no direct connectivity between the IP phones, Unified CM and outside the world." They also discussed theft of service, involving using network resources to place long distance calls that incurred high told charge or exploiting the resources by inter-trunk transfer can also occur elsewhere in the book.
Two Types of SIP Offer Invites
with SDP-Session Description Protocol
- Early Offer-Fast Start Invite
- SDP is sent with the Invite (advertises its CODEC/media capabilities,
encryption and other terms of call)
- Delayed Offer-Slow Start
Invite - Invite is sent without the SDP (called party advertises CODEC/media,
etc.) The "Offer" typically defines the media characteristics supported by the
device (media streams, CODECs, directional attributes, IP address, and ports to
use). The Offer Invite is contained in
the Session Description Protocol fields sent in the body of a SIP signaling
message. The SIP endpoint receiving the Offer sends an "Answer" in the SDP
fields of its SIP response, with its corresponding matching media streams and
codec, whether accepted or not, and the IP address and port on which it wants
to receive the media streams. Details on SDPs can be found in RFC-3261. Note: In either case, codec/media selection by
either called party or calling part in not unilateral decision but rather a
negotiation.
If an MTP or SBC is involved in
the either invite process, they can also act as "proxy servers" to negotiate
terms of either fast or slow start invitations.
Some reasons for using Early
Media include:
· The called device might want
to establish an Early Media RTP path to reduce the effects of audio cut-through
delay (clipping) for calls experiencing long signaling delays or to provide a
network-based voice message to the caller.
· The calling device might want
to establish an Early Media RTP path to access a DTMF-Dual Tone Multi-Frequency
or voice-driven IVR-Integrated-Interactive Voice Response system.
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For SIP Course details click here.
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Book Review
"VoIP Performance Management and Optimization"
By Adeel
Ahmed, Habib Madani,
Talal
Siddiqui.
For Cisco Press details click here.
There is really nothing about
this book I don't like except that it is only on paper, not electronic but read below and you get access to it online. That
is the only negative think I can say about this book and if you are serious
about VoIP (SIP) performance, QoS, security, monitoring and infrastructure
integration (hardware) and more, then you need to read and know everything that
is in this book. And like what my mom
said to me a long time ago, "if you think you know what's going on, then
you are really full of s##." Seriously there is just too much really
good information to mention in less than 300 words (trying to keep in
brief). Here's one of my favorites: on
page 262 "signaling traffic is also vulnerable to attack, including Spam over Internet
Telephony-SPIT. SPIT leverages SIP proxy
impersonation to sent unsolicited bulk messages to SIP endpoints. VoIP phishing (vishing or fishing) involves
CallerID spoofing and then call rerouting to dummy IVR systems for further exploitation
of the SIP call processing resources." This one of the many great "actionable" tutorials you will find in the
book.
With your book
purchase you are entitled to free, instant online access to that book on Safari
Books Online for 45 days. After you've completed your purchase, you will
receive instructions on how to log into Safari Books Online. If you do not want
to receive online access to the book, simply uncheck the box for Instant Online
Access in your cart.
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