© Copyright AARNet Pty Ltd Development of SIP-H.323 Gateway Project Ruston Hutchens 20 th APAN...

©Copyright AARNet Pty Ltd

Development of SIP-H.323 Gateway ProjectRuston Hutchens

20th APAN Meeting, Taipei, Taiwan

Thursday 25rd August 2005

v2


SIP-H.323 Gateway project

• Motivation–Large deployment base of H.323 terminals (over 2.9

million calls placed over the AARNet H.323 VoIP network in 2004)

–Will SIP replace H.323? Will H.323 die instantly? Will these protocols run in parallel?

• As a pre-requisite to using SIP, we need either an endpoint that has a SIP protocol stack, or

• a means to translate the SIP messages to the protocol that is natively supported


SIP and H.323 comparisons

SIP H.323

"New World" - a relative of Internet protocols - simple, open and horizontal

"Old World" - complex, deterministic and vertical

IETF ITU

Carrier-class solution addressing the wide area

Borne of the LAN - focusing on enterprise conferencing priorities

Taken from www.sipcenter.com



SIP H.323

A simple toolkit upon which smart clients and applications can be built. It re-uses Net elements (URLs, MIME and DNS)

H.323 specifies everything including the codec for the media and how you carry the packets in RTP

Leaves issues of reliability to underlying network

Assumes fallibility of network - an unnecessary overhead

SIP messages are formatted as text.

H.323 messages are ASN.1 binary encoded, adding complexity




SIP H.323

Minimal delay - simplified signaling scheme makes it faster

Possibilities of delay (up to 7 or 8 seconds!)

Slim and Pragmatic The suite is too cumbersome to deploy easily

Seamless interaction with other media - services are only limited by the developers imagination

Services are nailed-down and constricted

Many vendors developing products

The majority of existing IP telephony products rely on the H.323 suite



Ways to achieve protocol conversion

We have used three products to convert SIP to H.323 and vice-versa

Asterisk – Open source PBX (http://www.asterisk.org/)

Cisco Multiservice IP-to-IP Gateway (http://www.cisco.com)

“Back to Back” gateway (AS5300 with 2xE1)

REMEMBER: We’re only converting the call signalling, RTP is the same. (media transcoding is not required)

*Other products becoming available eg: Yatehttp://yate.null.ro/ - Yet another Telephony engine

http://www.asterisk.org/

http://www.cisco.com/

http://yate.null.ro/


H.323 protocol stack – see, it’s confusing

Source: www.protocols.com


H.323 protocol flow – with gatekeeper



H.225 Protocol

• H.225.0 is a standard which covers narrow-band visual telephone services defined in H.200/AV.120-Series Recommendations. It specifically deals with those situations where the transmission path includes one or more packet based networks.

• H.225.0 describes how audio, video, data and control information on a packet based network can be managed to provide conversational services in H.323 equipment.

• H.225 messages include

– ALERTINGCALL PROCEEDINGCONNECTCONNECT KNOWLEDGEPROGRESSSETUPSETUP ACKNOWLEDGE



H.245 Protocol

• H.245 is line transmission of non-telephone signals. It includes receiving and transmitting capabilities as well as mode preference from the receiving end, logical channel signalling, and Control and Indication. Acknowledged signalling procedures are specified to ensure reliable audiovisual and data communication.

• H.245 messages are in ASN.1 syntax. They consist of an exchange of messages. MultimediaSystemControlMessage message types can be defined as request, response, command and indication messages. The following lists SOME of the additional message sets are available:

• Master Slave Determination messages • Terminal capability messages • Logical channel signalling messages • Round Trip Delay messages • Communication Mode Messages • TerminalID • Commands and Indications



RAS Protocol

• The Registration, Admission and Status (RAS) channel is used to carry messages used in the gatekeeper discovery and endpoint registration processes which associate an endpoint's alias address with its call signalling channel transport address. The RAS channel is an unreliable channel. Since the RAS messages are transmitted on an unreliable channel, H.225.0 recommends time-outs and retry counts for various messages. An endpoint or gatekeeper which cannot respond to a request within the specified timeout may use the Request in Progress (RIP) message to indicate that it is still processing the request. An endpoint or gatekeeper receiving the RIP resets its timeout timer and retry counter.

• Source: www.protocols.com


SIP Protocol FlowSIP UA

INVITE (with SDP)100 Trying

407 Auth Req

ACK

INVITE With auth100 Trying

180 Ringing200 OK(with SDP)ACK

BYE

RTP

200 OK


SIP to H.323 Protocol Flow

SIP UA IP-IP Gateway H.323

INVITE (with sdp)100 Trying

Call Proceeding

Alerting Connecting

Open logical channel

Open logical channel inc TCS

BYE Close logical channel ACK

180 Ringing

200 OK (with SDP) ACK

RTP

Close logical channel

ACK


H.323 to SIP Protocol Flow – note delayed SDPH.323 Endpoint IP-IP Gateway SIP UA

Setup

INVITE (no sdp)100 Trying

180 RingingAlerting Connecting 200 OK

(with SDP)Term capability setMaster/slave det.

Terminal Cap Set + ACK, MS ACKTerm capability ACK

Master/slave ACKOpen logical channel

Open logical channel ACK

ACK with SDP

BYE

Close logical channel

Close logical channel ACK

Call Proceeding

RTP


Cisco

• SIP-H.323 and H.323-SIP support was introduced in IOS release 12.3(11)T

• Supports Voice calls. Video is supported for some calls

• Not a big performance impact, as no media transcoding is involved. At worst case, just copy the media stream from input buffer to output buffer

• Costs money (2 cisco routers, 1or 2 ‘expensive’ software license)


Back to back gateway

• Achieved through the use of two E1 ports on a gateway

• ISDN cross-over cable is connected between the two

• Requires 2x media encode-decode

• Increases latency

• Relatively easy to setup


Asterisk

• Asterisk “likes” media to flow-through, rather than flow-around.

• Supports Voice and Video

• Asterisk must know about the RTP protocol before it will allow calls to be setup

• H.323 is not a supported channel type “out of the box”. You need to compile a separate channel driver – version-ing IS a problem

• Free


SIP-H.323 Gateway Network Architecture (Cisco IP-IP Gateway)


SIP-H.323 Gateway Network Architecture (Asterisk Server)


Codec Negotiation

• Currently both IP-IP gateway and Asterisk products have issues selecting the correct codec

• Back to back gateway does not have this issue, as it converts to ISDN first

• IP-IP Gateway only “supports” H.323 fast-start


Dial plan issues

• Desire for “A” party not required to be aware of “B” party’s technology

• Other option to have an “access code” – this requires either the user to know the B party’s technology type, or the SIP proxy or gatekeeper to insert a routing prefix


Product comparisons

• Asterisk: Was never designed to do what we are asking of it. It was designed as a soft-PABX, with small-medium number of telephones attached, and small numbers of trunk connections

• H.323 support is an “add on”, and is not supported all that well.

• Back-to-back is not elegant, with some latency introduced, and possible voice quality degradation


Product Comparisons

• Cisco IP-IP Gateway/Gatekeeper• Was originally designed as a H.323 – H.323 product. • Has very good H.323 support• Was not designed as a softswitch, so does not support

large “routing tables”. Routing is achieved by adding dial-peers. Each dial peer requires approximately 6kb of memory.

• While this allows 5000 theoretical dial peers on an “average” router, this becomes unmanageable


SIP and H.323 Feature compatibility

• Obviously, some features exist only in one protocol, but not in the other. Regardless of how good the protocol converter is, if features don’t exist in the protocol standard, they can’t be supported)

• Examples include:– Security– Presence– Support for different types of sessions (IM)


Support

• Back to back gateway can use almost any device that has 2 or more E1/T1 ports, and supports sip calls and H.323 calls. In our case, we used a Cisco AS5300, which is supported by Cisco

• Cisco IP-IP Gateway is supported by Cisco

• Asterisk support can be purchased from the creator, Digium Inc, as well as other 3rd parties.


Acknowledgements

• We wish to acknowledge the assistance of Cisco Systems with assistance configuring and fault-finding the IP-IP gateway/gatekeeper

• We wish to acknowledge the assistance of Broadreach Services +61 2 8270 1000 for some assistance configuring the Asterisk product


Other Products - YATE

YATE is an easily extensible, next-generation telephony engine, currently focused on VoIP. Yate is able to support Voice, Video and IM. The software is written in C++ and it supports scripting in various programming languages

Yate can be used as:

•VoIP server •VoIP client •VoIP to PSTN gateway •PC2Phone and Phone2PC gateway •H.323 gatekeeper •H.323 multiple endpoint server •SIP session border controller

•SIP router •SIP registration server •IAX server and client •IP Telephony server and client •Call center server •IVR engine •Prepaid and postpaid cards

system


YATE as an H.323 to SIP gateway

• Requires Open H.323 and pwlib (versioning issues)

• Does not need to understand the codec (c/f Asterisk)

• First we have to set up the H.323 channel to use RTP pass-trough mode. In this channel the RTP mode must be configured globally, not per call. In the file h323chan.conf we put:

• [general] external_rtp=yes • passtrough_rtp=yes • [codecs] default=no • mulaw=yes • alaw=yes • g723=fake • [ep] • faststart=on

• h245tunneling=off


YATE client

• YATE client is a specialised

form of YATE.

Simple interface

Currently for GNU/Linux

Windows (requires GTK2)


Ethereal: Voip call analysis


Opportunities for use within the APAN community

• This could be used to peer different country’s dissimilar Voice (and Video) over IP networks. Some results for Research Networks, and their members could be:

–Cheaper telephone calls–Easier video conferencing for users–Better collaboration

• If you’re interested, please contact the presenter, or attend the APAN SIP-H.323 working group meeting On Friday, at 14:00 in room H

www.aarnet.edu.au


IP to IP Gateway/Gatekeeper – Other uses

• What are the uses? –Peering two H.323 networks

• Network demarcation (hide internals of network)

• Dial plan translation• CDR billing record collection point


IP to IP Gateway/Gatekeeper – other uses

• What are the uses? (cont)–Gateway to Internet for IP Phones

• Can act as a “firewall” for IP phones

Eg. All calls to Internet destinations are made through the IP-IP gateway, so the phone does not need to be visible from the Internet

• Present a single IP address to the outside world

• Demarcation point for collecting CDRs from phones


IP-IP International Gateway/Gatekeeper GDS peering configuration


www.aarnet.edu.au

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