These are partly notes I have jotted down from what I am reading and also links to helpful VOIP sites. I lay no claims to ownership, except where indicated.

Sunday, January 24, 2010

Today's jottings


Actual voice data is transported using media transmission protocols like RTP and RTCP.


- defines format for delivering audio and video over the Internt
- delivers audio and video streams over a network


- provides OOB control info for an RTP flow.

2 Protocols which enhances the use of RTP:

- cRTP which compresses IP/UDP/RTP headers on low speed serial links.

- SRTP which provides encryption, message auth and integrity, and replay protection to RTP data.

RTP and RTCP are both built on UDP.


- e2e network transport functions like payload-type ID, sequence numbering, time stamping, delivery monitoring.

- runs on top of UDP in order to use UDP's multiplexing and checksum services. This is because it has no standard TCP or UDP port on which it communicates (though it is sometimes config to use ports 16384 or 32767). UDP comms must be done via an even port and the next higher odd port is used for RTCP.

- can carry any data with real-time characteristics. Call setup and teardown is usually performed by SIP. It is difficult for RTP to traverse firewalls since it uses dynamic port config. It works well with queuing to prioritize voice traffic over other traffic.

- designed for multicast sessions. It defines roles os sender/receiver and also roles of translator/mixer to support multicast reqs. But it is often used for unicast sessions.

- includes various services like payload-type ID, sequence numbering, time stamping, delivery monitoring.

RTP is used:

- with RTSP (real time streaming protocol) in streaming media systems
- wih H.323 or SIP in video conferencing and push-to-talk systems

What makes RTP so critical a component for VOIP is the fact that it enables destination devices to reorder and retime voice packets before they are played out to users.

An RTP header contains a time stamp and a sequence number, which allows the receiving device to buffer to remove jitter and latency by synchronising the packets to play back a continuous stream of sound.

Sequence number is only used for ordering the packets. There is no request by RTP for retransmission if a packet is lost.


-sender report packet
-receiver report packet
-source description RTCP pkt
-goodbye RTCP pkt
-app-specific RTCP pkt


-provides oob control info for an RTP flow

it does not transport any data itslef, but provides feedback on the QoS being provided by RTP.

-used for QoS reporting

Monitors quality of data distribution and provides control info

-provides mech for hosts involved in an RTP session
-provides a time stamp based on data pkt sampling
-provides separate flow from RTP for transport use by UDP


RTP is divided into 2 portions, data and header.

The data portion is a thin protocol and provides support for real-time properties of applications, such as continuous media, including timing reconstruction, loss detection and content identification.

The header is made up of iP/UDP/RTP segments which is 20/8/12 bytes which equals 40 bytes. This is simply too large and it is inefficient to send this without compressing it. The header compression feature compresses the 40 bytes to approx 2 to 4 bytes.

RTP payload is 20 to 150 bytes for audio apps that use compressed payloads.

RTP header compression should not be used on any high-speed interfaces - anything over T1 speed.

SRTP - secure RTP standardizes the utilization of only a single cipher, AES. which can be used in 2 cipher modes and turns the original AES cipher block into a stream cipher.

The two cipher modes:

1. Segmented integer counter (SIC) mode: allows random access to any block and is essential for RTP traffic running over an unreliable network with possible loss of pkts. AES running in this mode is the default encryption algorithm, with a default encrytion key length of 128 bits and a default session salt key of 112 bits.

2. f8 mode: a variation of output feedback mode. Default values above applies.

SRTP also allows users to disable the cipher using 'NULL cipher'.


Gateways provide a way for IP telephony networks to connect to the pSTN, PBX, etc.

A voice gateway functions as a translator between different network. Coverts protocols between terminals such as H.323 and PBX. Connects company networks to PSTN, PBX, etc.



Two subcategories here:


Connects IP telephony ntwk to POTS.
Provides FXS ports to connect analog tels, interactive voice response (IVR)systems, fax machs, PBX systems and voice-mail systems.


Connects IP telephony ntwk to the PSTN central office (CO)or a PBX.
Provide FXO ports for the PSTN or PBX access, and E&M ports for analog trunk connection to legacy PBX.
Analog direct inward dialing [DID] is also available for PSTN connectivity.


Cisco access digital trunk gateways connect an IP telephony ntwk to the PSTN or PBX via digital trunks like PRI common channel signaling (CCS), BRI and T1 or E1 channel associated signaling (CAS)


-support gateway protocols
-provide advanced gateway functionality
-work with redundant CUCM
-enable call survivability
-provide Q signaling [QSIG]support
-provide fax and modem support

MGCP might be the preferred option to H.323 for gateway config due to its simpler config. It works well with older Cisco IOS versions because it supports call survivability during a CUCM failover from a primary to a secondary CUCM.

H.323 on the other hand may be preferred for its advanced features.


-standard for integrating voice-mail systems to PBX or Centrex systems.
-connecting to a voice-mail system via smdi using either analog fxs or T1 PRi requires either SCCP or MGCP since H.323 does not identify the specific line that is being used by a group of ports.


1. Dual Tone Multifrequency (DTMF) relay capabilities:

-each digit dialed with tone dialing is assigned a unique pair of frequencies
-DTMF tones are separated from the voice bearer stream and sent as signaling indications via the gateway protocol.

2. Supplementary services support:

-provide user functions such as hold, transfer and conferencing.

3. Must have the ability to REHOME to a secondary CUCM in the event that the primary fails.

4. Must enable call survivability in the CUCM.

- the voice-gateway preserves the RTP bearer stream (the voice conversation) between two IP endpoints when the CUCM to which the endpoints are registered is no longer accessible.

5. Should provide QSIG support (which is becoming the standard in Europe and North America)

- with QSIG, the cisco voice pkt appears to PBXs as a distributed transit PBX that can establish calls to any PBX or other telephony endpoint served by a Cisco gateway, including non-QSIG gateways.

6. Should also provide fax and modem support

- the fax image is converted from an analog signal and is transmitted as digital data over the pkt ntwk.

Gateways are normally deployed as edge devices on a network.

Some enterprise gateway hardware:

1. Cisco 2800 Series Integrated Services Router: four models-

-Cisco 2801
-Cisco 2811
-Cisco 2821
-Cisco 2851

-provides up to five times the overall performance
-up to 10 times the security and voice performance
-embedded service options
-dramatically increased slot performance and density

-can deliver simultaneous, high-quality, wire-speed services up to multiple T1/E1 or xDSL connections.

The routers offer:

-embedded encryption acceleration
-voice digital signal processor(DSP) slots

-intrusion prevention systems (IPS) and firewall functions

-optional integrated call processing and voice-mail support

-high-density interfaces for wide range of wired and wireless connectivity requirements.

-sufficient performance and slot density for future network expansion reqs and advanced apps.

2. Cisco 3800 Series Integrated Services Router

also feature:

-embedded security processing
-significant performance and memory enhancements
-new high-density interfaces which deliver the performance, availability and reliability reqd to scale mission critical security, IP telephony, business video, network analysis and web apps in the most demanding enterprise environments
-deliver multiple concurrent services at wire-speed T3/E3 rates

-are designed to embed and integrate security and voice processing with advanced wired and wireless services for rapid deployment of new apps

-support bandwidth reqs for multiple FE interfaces per slot, TDM interconnections and fully integrated power dist to modules supporting 802.3af PoE.

-supports the existing portolio of modular interfaces.

3. Cisco Catalyst 6500 Series Switches

-high-performance and feature rich platforms that can be used for voice gtws by installing a Cisco Communications Media Module (CMM)

-CMM is a line card for the Cat 6500 series. It provides flexible and high-density VoIP gtwy and media services.

-The Series can handle many digital trunk interfaces eg, up to 144 T1/E1 conns by using 8 CMMs with 18 ports each.


1. Cisco 1751-V Modular Access Router

-supports multiservice integration of video, voice, data and fax traffic.
-offers many WAN-acces and voice-interface options, VoIP high-performance routing with bandwidth management, inter-VLAN routing, and VPN access with a firewall options.

2. Cisco 1760-V Modular Access Router

-19 inch rack mount solution geared towards small-medium businesses
-supports multiservice integration of video, voice, data and fax traffic.
-offers many WAN-acces and voice-interface options, VoIP high-performance routing with bandwidth management, inter-VLAN routing, and VPN access with a firewall options.

3. Cisco 2600XM Series Multiservice Routers
4. Cisco 3600 Series Multiservice Routers
5. Cisco 3700 Series Multiservice Routers

All of them are either EOS or almost-EOS.


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