Why Choose SIP?
SIP is independent of the media used, which allows flexibility to initiate sessions for different media types.
Intelligence is distributed to endpoints, not to a single call-control component.
IETF derived SIP from HTTP, so it is easy for third-party developers to create applications for it.
SIP provides graceful support of protocol extensions.
SIP is independent of any security protocol.
SIP is a peer-to-peer protocol not an IP-to-PSTN gateway control protocol like MGCP.
SIP runs on top of several transport protocols, including UDP, TCP, and SCTP.
SIP is a peer-to-peer, multimedia signaling protocol that integrates with other Internet services, such as e-mail, Web browsers, voice mail, instant messaging, multiparty conferencing, and multimedia collaboration. When used with an IP infrastructure, SIP helps to enable rich communications with numerous multivendor devices and media. SIP can set up individual voice or conference calls, videoconferences and point-to-point video-enabled calls, Web collaboration and chat sessions, or instant messaging sessions between any number of SIP-enabled endpoints including IP phones, PCs, laptops, personal digital assistants (PDAs), and cell phones.
SIP is attractive as a signaling standard because it can connect and control communication sessions between applications, independent of media type or the function performed by the end applications. SIP is known as a methods-based signaling protocol because it provides the methods to connect, signal, and control sessions. In that sense, SIP is quite different from a
functionally based signaling protocol, such as QSIG, which is used not only to establish sessions, but also to define the specific features those sessions can support.
The distinction between SIP and a functionally based signaling protocol is important because it greatly affects interoperability and flexibility. As a peer-to-peer protocol, the intelligence involved in SIP-enabled applications is distributed to endpoints and other components, not centralized in a single call-control component. New features can be added without upgrading infrastructure components such as proxy servers.
SIP is based on HTTP, so developers who are typically knowledgeable about HTTP and Web-based applications do not require intimate knowledge of the SIP infrastructure in order to write SIP-enabled applications. The common connection to HTTP opens up the application
development process to third-party developers who can create targeted, vertically oriented applications. For example, internal users at a financial services company are likely to want features that are considerably different from those used by telemarketers in an outbound call center. SIP enables independent software vendors that have expertise in each market to develop applications specific to those areas. This type of open development environment represents a dramatic shift from the traditional TDM-based PBX paradigm discussed above. By opening up the application development process to more players, SIP promises to provide more innovation in less time and at less cost.
SIP also gracefully supports protocol extensions, so that applications can support advanced features and can still interoperate with other, less functional applications. Consider the following example. Three colleagues are on a conference call. Two of them are at a
headquarters location where their SIP-enabled IP phones support video capabilities. The third is at a remote office that does not support video phones. SIP establishes the conference among the three users and enables the video portion for the two users whose equipment supports it. The third user participates in a traditional audio call. This is a shift from the least-common denominator approach in which only functions supported by all users are implemented (none of the colleagues would be able to use video).
In this way, SIP supports innovation within applications that can be combined, yet still works with similar applications that support features that have not been extended. In fact, SIP extensions define how SIP discovers the feature set each endpoint supports, and how SIP establishes each call accordingly.
Summary
This topic summarizes the key points discussed in this lesson.
© 2005 Cisco Systems, Inc. All rights reserved. GWGK v1.01-29
Summary
An H.323 gateway performs the following services in a VoIP network:
Translation between audio, video, and data formats
Conversion between call setup signals and procedures
Conversion between communication control signals and procedures
The H.323 setup exchange using the Fast Connect abbreviated procedure reduces the number of round-trip exchanges by performing the capability exchange and logical channel assignments in one round trip.
© 2005 Cisco Systems, Inc. All rights reserved. GWGK v1.01-30
Summary (Cont.)
DTMF relay solves the problem of DTMF distortion by transporting DTMF tones out-of-band or in-band in the RTP stream with different coding from the voice.
SIP DTMF considerations include the need to think about out-of-band to in-band conversion.
H.323 performs well in a TDM environment making it a good choice for migration.
MGCP offers the advantage of powerful and scalable central call control.
SIP is less mature than either H.323 or MGCP, but it offers developers the opportunity to build applications for SIP networks more easily than in other networks.
Lesson Self-Check
Use the questions here to review what you learned in this lesson. The correct answers and solutions are found in the Lesson Self-Check Answer Key.
Q1) Why is H.323 is considered to be an umbrella protocol? (Source: Overview of H.323 Gateways)
Q2) How does H.323 interoperate with legacy voice networks? (Source: Overview of H.323 Gateways)
A) H.323 is based on the H.225 standard, which interoperates with existing voice networks.
B) H.323 uses the ISDN Q.931 protocol to interoperate with legacy voice networks.
C) H.323 is based on the H.245 standard, which interoperates with existing voice networks.
D) H.323 uses the EGIE, Phase 2 to interoperate with legacy voice networks.
Q3) Which three services are performed by a H.323 gateway? (Choose three.) (Source:
Overview of H.323 Gateways)
A) translation between audio, video, and data formats
B) registry maintenance of devices in the multimedia network C) conversion between call setup signals and procedures
D) conversion between communication control signals and procedures E) bandwidth management and alternate gateway locating
F) address translation and control access to legacy networks G) termination of all ISDN PRI Layer 2 (Q.921) signaling functions
Q4) What does the Fast Connect procedure do in one round trip between H.323 gateways?
(Source: H.323 Call Flow)
A) establishes the capability exchange and logical channel assignments B) defaults the H.252 gateways to H.323 version 3 for all calls C) establishes rate-limit traffic by precedence
D) establishes the capability exchange and codec assignments
Q5) What are three of the benefits of the Cisco H.323 gateway? (Choose three.) (Source:
Choosing a Gateway Protocol?)
A) designed as a generic transaction protocol for session initiation B) works well for organizations with centralized management and control C) gateway resource availability reporting
D) shorter voice cut-through times
E) gateway fallback support Cisco CallManager F) allows TDY users to communicate with voice users G) gateway support for DTMF digit relay
Q6) What two conditions must be met for a Cisco H.323 gateway to be able to send digits out-of-band? (Choose two.) (Source: H.323 DTMF Relay Considerations)
A) A Cisco H.323 gateway has enabled the chosen method of DTMF relay under dial-peer configuration.
B) A Cisco H.323 gateway has enabled the chosen method of DTMF relay under global configuration.
C) The global dial peer must be configured so it is capable of receiving the forwarded digits from the remote end.
D) The dial peer must be configured so it is capable of receiving the DTMF option being used.
Q7) Briefly describe how MGCP works. (Source: Overview of MGCP Gateways)
Q8) What is an example of an MGCP call agent? (Source: Overview of MGCP Gateways) A) a H.323 gateway with MGCP enabled
B) the Cisco CallManager
C) a phone set up as an ACD (IPCC Express) phone agent
D) a desktop set up as an ACD (IPCC Express) agent (not Supervisor)
Q9) How is voice data transmitted using MGCP? (Source: Choosing a Gateway Protocol) A) Voice data is transmitted using the RTP portion of the MGCP protocol.
B) Voice data is not transmitted via MGCP, only fax and modem tones are transmitted via MGCP.
C) Voice data is transmitted using the SIP portion of the MGCP protocol.
D) No voice data is transmitted through the MGCP protocol itself.
Q10) Is IETF support for the MGCP protocol a benefit? Briefly explain why or why not.
(Source: Choosing a Gateway Protocol)
Q11) Why is it a benefit to transport DTMF tones out-of-band? (Source: DTMF Relay Considerations)
A) The problem of DTMF distortion is solved.
B) Fax tones are transported via TCP rather than as voice over UDP or RTP, which makes the transport of these tones very reliable.
C) Dialed digits are transported in a separate carrier, like the D-channel of ISDN, thus conserving voice bandwidth.
D) Dialed digits can code the A, B, C, and D tones provided for in the DTMF standard.
Q12) Explain the function of each type of peer in the SIP protocol. (Source: Overview of SIP Gateways)
Q13) What are the two types of SIP servers? (Choose two.) (Source: Overview of SIP Gateways)
A) reversion server B) user agent server C) proxy server D) redirect server E) unified agent server F) universal proxy server G) registrar server H) registration server I) location server
Q14) What kind of transaction model is SIP based on? (Source: SIP Call Flow) A) a client-server model
B) an HTTP-like request and response transaction model C) a primary agentsecondary agent model
D) a peer-to-peer transaction model
Q15) Briefly explain what ensures SIP interoperability with H.323 and MGCP. (Source:
Choosing a Gateway Protocol)
Q16) What does the out-of-band to in-band DTMF relay for Cisco IOS voice gateways feature enable? (Source: SIP DTMF Relay Considerations)
A) support for the conversion of DTMF tones to counter the distortion found in voice-codes transporting dialed digits
B) DTMF-relay communication between SIP devices and non-SIP endpoints using Cisco CallManager
C) transcoding out-of-band tones into non G.711 audio codecs D) DTMF-relay to fax-relay communications between endpoints
Q17) In the Call Control Protocol column in the table below, write down the call control protocol (H.323, MGCP, or SIP) that you would choose for each gateway. The
following information may help influence your selections: All PBX connections are T1 PRI-enabled and QSIG-enabled except Gateway 1, which connects with T1 CAS.
SRST is enabled on Gateways 1, 4, 7, and 9. Cisco CallManager has not been deployed in Sao Paulo or Dallas. (Source: Choosing a Gateway Protocol)
© 2005 Cisco Systems, Inc. All rights reserved. GWGK v1.01-28
Choosing a Gateway Protocol
Span Engineering Voice Gateway Topology End of Phase 2 IP Telephony Migration
LD
Gateway Information
Gateway
Number Gateway Type Gateway Connects To Description Call Control Protocol 1 2 x VWIC-2MFT-T1 in a 2811 Nortel Meridian Opt 11c 4 x T1 CAS
2 1 x WS-6608-T1
1 x WS-SVC-CMM-6T1
PSTN Long Distance
8 x T1 PRI 5 x T1 PRI 3 2 x VWIC-2MFT-T1 in a 3845
2 x NM-HDV-2T1-48 in a 3845
Avaya MV1.3 PBX Nortel Meridian 1
3 x T1 PRI 4 x T1 PRI
4 2 x NM-HDV-2T1-48 in a 3845 PSTN 3x T1 PRI
5 2 x NM-HDV-2T1-48 in a 3845 Avaya Definity G3si 3 x T1 PRI
6 NM-HDV-2E1-60in a 3845 PTT
PISN
1 x E1 R2 1 x E1 PRI
7 VIC2-2BRI-NT/TE in a 2811 PSTN 2 x BRI
8 4 ports on WS-6608-T1 4 ports on WS-6608-T1
PSTN Long Distance
4 x T1 PRI 4 x T1 PRI
9 VWIC-1MFT-T1 in a 2821 PSTN 1 x T1 PRI
Lesson Self-Check Answer Key
Q1) H.323 is considered an umbrella protocol because it defines all aspects of call transmission, including call establishment, capabilities exchange, and network resource availability
Q2) B
Q3) A, C, D
Q4) A
Q5) A, C, G
Q6) A, D
Q7) MGCP is defined under RFC 2705. It is a plain text protocol that uses a master-slave relationship to fully control a gateway and its associated ports. The plain-text commands are sent to gateways, from call agents using UDP port 2427. Port 2727 is used to send messages from the gateways to the call agent.
Q8) B
Q9) C
Q10) Yes, it can be a benefit. Without being subject to the rigors of the ITU-T procedures and policies, the IETF can respond quickly to user demands, although the solutions can be less mature than those created by the ITU-T.
Q11) A
Q12) The user agent client is a client application that initiates a SIP request and the user agent server is a server application that contacts the user when a SIP invitation is received and then returns a response on behalf of the user to the invitation originator.
Q13) B, D
Q14) B
Q15) Although SIP messages are not directly compatible with H.323 and MGCP, these protocols can coexist in the same packet telephony network if a device that supports the interoperability is available.
Q16) B
Q17) H.323, MGCP, H.323, H.323, H.323, H.323, H.323 or SIP, MGCP, H.323 or SIP