Snom 4S Media Server User manual

snom 4S
User Manual
SIP Registrar/Proxy 2.12

snom technology AG • 3
snom 4S SIP Registrar Proxy 2.12
User Manual

snom technology AG • 3
snom 4S SIP Registrar Proxy 2.12
User Manual
snom 4S
SIP Registrar/Proxy 2.12

snom technology AG • 5
snom 4S SIP Registrar Proxy 2.12
User Manual
snom 4S Registrar Proxy Version 2.12 User Manual
3. Edition 2002
© 2002 snom technology Aktiengesellschaft. All Rights Reserved.
This document is supplied by snom technology AG for information purposes only to licensed users
of the snom 4S registrar proxy and is supplied on an “AS IS” basis, that is, without any warranties
whatsoever, express or implied.
Information in this document is subject to change without notice and does not represent any
commitment on the part of snom technology AG. The software described in this document is
furnished under a license agreement and may be used only in accordance with the terms of that
license agreement. It is against the law to copy or use this software except as specically allowed in
the license. No part of this document may be reproduced, republished or retransmitted in any form
or by any means whatsoever, whether electronically or mechanically, including, but not limited to,
by way of photocopying, recording, information recording or through retrieval systems, without the
express written permission of snom technology AG.

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snom 4S SIP Registrar Proxy 2.12
User Manual
Welcome to snom 4S !
SIP is becoming more and more accepted in the area of VoIP. Many companies
are working on SIP solutions and making great products that will make telephony much
easier and better. However, a easy to use and affordable SIP proxy is hard to get at the
moment. But SIP telephony without a proxy makes no sense. That is why we are making
this simple and straightforward proxy available.
snom 4S stands for “snom soft switch for small and medium enterprises“.
That means, the snom 4S proxy was designed for environments handling up to 1000
users with normal trafc. In environments where you need more features and better
scalability, we would be happy to refer you to other companies offering carrier grade and
feature-rich proxies that solve these problems.
This product is a proxy/registrar which means this software is responsible
for locating users. Features like Follow me and group calling are therefore supported;
however media services like mailbox and music on hold are not part of the software. You
should use a media server for this.
Interoperability is important to us. We have tried to stick to the SIP standard
as well as possible and tested the phones of other vendors. We hope that this will help
to build up a ourishing VoIP telephone industry in which the products of the different
vendors work together like the products in the computer industry do today. We believe
that having a choice is good for you and therefore good for us.
This manual gives you a brief introduccion to VoiP and SIP, explains the
installation process for Windows and Linux and shows how to run the SIP proxy. For
additional snom 4S information, please visit our Web site at http://www.snom.de and
if you have any comments and suggestions about snom 4S, please contact us through
snom technology AG’s support link Web site. We would appreciate your feedback.
We hope that this SIP proxy helps you get VoIP up and running!
Thank you and....
have fun using the snom 4S!
Sincerely,
Dr. Christian Stredicke Nicolas Peter-Pohland
Managing Director Managing Director

6 • Table of Contents
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Table of Contents
Welcome to snom 4S ! ............................................ 5
Voice over Internet Protocol ................................ 11
Why SIP?............................................................................. 11
Open Standards ................................................................... 12
What You Can Expect and What Not ........................................ 12
The SIP Architecture ............................................ 15
User Agents ......................................................................... 15
Registrars ............................................................................ 15
Proxies................................................................................ 16
Media Server........................................................................ 16
Gateways ............................................................................ 16
The snom 4S Solution Framework ........................ 19
Proxy .................................................................................. 19
SIP NAT Gateway.................................................................. 20
Media Server........................................................................ 20
Installation .......................................................... 23
Windows ............................................................................. 23
Uninstalling in Windows ......................................................... 33
Linux .................................................................................. 34
Automatic starting under SuSe Linux ........................................................................ 34
RedHat........................................................................................................................................ 35
snom 100 Registration........................................................... 37
Microsoft Messenger Registration ............................................ 38
General Concepts ................................................. 45
Security ............................................................................. 45
Reliabilty ............................................................................. 46
State .................................................................................. 47
Overlap Dialling.................................................................... 48
Sequential Forking ................................................................ 49
Network Address Translation .................................................. 50
Routing ............................................................................... 51
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Conguration ....................................................... 53
Licensing ............................................................................. 53
Hostnames ............................................................................................................................... 54
IP Addresses............................................................................................................................ 55
License Key.............................................................................................................................. 55
General Settings................................................................... 55
SIP Port...................................................................................................................................... 56
Proxy realm ............................................................................................................................. 56
TCP threshold ......................................................................................................................... 57
Log Level................................................................................................................................... 57
HTTP port.................................................................................................................................. 57
HTTP User and Password .................................................................................................. 57
Registration ......................................................................... 58
Require Authorization......................................................................................................... 58
Trace REGISTER.................................................................................................................... 58
Save Registrations to File................................................................................................. 58
Min and Max registration time ....................................................................................... 59
Default Q................................................................................................................................... 59
Reject Registrations Across NAT................................................................................... 59
Routing ............................................................................... 60
Protected Destinations....................................................................................................... 60
Max Forwards ......................................................................................................................... 61
Call Log File ............................................................................................................................. 61
NAT Gateway .......................................................................................................................... 61
Do not signal loose routing.............................................................................................. 61
Do not Record-Route if Route is present.................................................................. 61
Remove Tags on 18x .......................................................................................................... 62
Sequential Forking Time ................................................................................................... 62
User Administration .............................................................. 62
Dial Plan.............................................................................. 64
How it works ........................................................................................................................... 64
Example 1: Setting up overlap dialling ..................................................................... 66
Example 2: North American Dial Plan........................................................................ 67
Example 3: Do not allow cell phone numbers to certain users .................... 67
Error-Information ................................................................. 68
Welcome Message................................................................. 71
DNS.................................................................................... 71
What is DNS?.......................................................................................................................... 71
Setup DNS................................................................................................................................ 72
Maintenance......................................................... 75

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Registered Users .................................................................. 75
Call Logs ............................................................................. 76
Condition for Logging a Call............................................................................................ 76
Call Logs in the Web Interface....................................................................................... 77
Call Log File Format............................................................................................................. 78
SIP Message Flow ................................................................. 78
Logging ............................................................................... 80
More Information ................................................. 83
Release Notes ...................................................................... 83
Standards............................................................................ 84
Other useful information ........................................................ 84
Footnotes ............................................................................ 84
Index ................................................................... 88

8 • Table of Contents

snom 4S • SIP
Registrar/Proxy 2.12
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snom 4S • SIP
Registrar/Proxy 2.12
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Voice over Internet Protocol
Today there is a wide choice of different switched network
products. Telephones have now been being built for more than a century,
and their technology is well-understood and proven. Why choose a different
technology?
Modern communication infrastructures transport much more than just
one application: email, http, les, instant messages, videos, music, so it is only
natural to include voice in the list of applications and use one infrastructure for
all of them.. Voice is a real time application. Sending voice over the Internet
Protocol is called “VoIP”. The delay between sending a packet to the network
and receiving it needs to be minimal and constant and this makes specic
demands on this application.
Most network equipment can already full this real time requirement.
Virtually all switches currently on sale support a VLAN with different priorities in
the network, and the vast majority of higher layer network equipment supports
some means of transporting packets with different qualities (DiffSrv). The LAN
usually supports a bandwidth of 100 MBit/s, which is more than enough to
allow voice to ow through the network, and adherence to a certain set of rules
ensures that this bandwidth is enough to supply superior telephone quality.
The Internet backbone’s ability to transport large loads is increasing on a daily
basis, and global communications are now ruled by the Internet.
Why SIP?
There has been a “protocol war” regarding the “best” way to set up
a phone call. In the mid 90s, H.323 was the rst attempt to unify the VoIP
industry under a common standard,and move the world of telephony into
the computer industry, using most of the methods known from ISDN. Seen
however, from today’s perspective, the resulting technology was far too
complex, so products based on this technology did not work well together. The
late introduction of “supplementary services” (H.450.x) not only introduced
another level of complexity, but was also simply too late.
By the late 90s, the Session Initial Protocol (SIP) had been proposed
(RFC 25431). SIP follows the paradigms of the Internet, and is built upon the
same principles used by http and email. Moreover, it has found an enthusiastic
community of researchers and developers who like the idea of applying
Internet technology to real time communications. More and more applications
are being put into SIP, telephony being just one of them.
So far more than 150 drafts have been proposed for extending the
SIP protocol. All kinds of solutions are being addressed in these documents,
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and the highly dynamic eld of this new real time communication technology is
resulting in evolutionary pressure to nd the best common denominator.
Most of the “big players” have jumped on the SIP train. Microsoft
Messenger is based on SIP and industry giant Cisco offers SIP extensions
to most of their products. International organizations like ETSI host SIP
interoperability events, and next generation mobile technology will be
integrated with, if not based on, SIP.
Open Standards
Open standards dene the rules of the game. Interoperability allows
customers to choose between the products of different vendors and opens
up competition below the system level. This can be advantageous for the
customer, as the computer hardware industry has shown.
Many vendors therefore advertise their usage of an “open standard”,
dening this term as “we make the way our standard works public”. However,
this cannot really be called “standard” if only one vendor is using it. The
disadvantage is that customers still have a limited choice of products they can
buy.
There is no one objective denition of an open standard. However,
something approaching an open standard could be reached if a signicant
number of vendors offered products using the same standard, giving
customers the possibility of combining products to create a system. SIP is just
such a standard.
What You Can Expect and What Not
Telephony is more than making calls from A to B. SIP supports all
kinds of transfers, call parking and call picking, user searches (Follow-me),
mailbox support, and all the other features known from traditional telephony.
In addition to this, telephones can now indicate their willingness to receive
calls and the probability of nding a specic user.
You can call a PSTN number from a SIP phone just as you did ten
years ago. The network will usually be set up to terminate these calls on a
gateway which translates the packet stream into a switched network signal.
You can also dial email-like numbers like “sip:fred.[email protected]
”, and you can reach your sales team under the same telephone number and
email address.
Internet telephony is still a “best effort” communications technology

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and does not always necessarily support the quality of transport telephony
requires. If you are placing a phone call over the public Internet, there is no
guarantee that a packet will be transported within a reasonable time. Usually
there is acceptable quality, but it may happen that calls suddenly break off, that
there is signicant delay, or that packet loss causes stuttering. It is important
that users know what to expect: Cell phone users know that driving through
a tunnel may break the call, and Internet telephony users must be aware that
talking for free may compromise call quality.

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Registrar/Proxy 2.12
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The SIP Architecture
User Agents
In a SIP network, the phones2 make up most of the brain power,
unlike traditional telecoms equipment which can not scale so well:
• they play and record audio,
• they compress and uncompress the digital audio,
• they do echo compensation,
• they compensate for packet jitter and packet loss,
• they look for the destination,
• they retrieve their conguration information,
• they keep track of phones that offer a call pickup,
• they publish their state upon request,
• they determine and publish the probability of nding somebody,
• they terminate one or more identities,
• they redirect calls when nobody picks up,
• they are part of a virtual LAN,
• they search address books (LDAP),
• they search internet addresses (DNS A, DNS SRV),
• they usually include a web server,
• they send an receive instant messaging information,
• they publish network management information (SNMP),
• they behave like normal computers on the network (DHCP, DNS).
Phones are also called “user agents” and behave in a client/server
manner (somebody being the user agent client, UAC and somebody the user
agent server, UAS). In SIP, there is no conceptual difference between a hard
phone and a soft phone. The snom 100 VoIP phone or Microsoft Messenger are
examples of this kind of system.
Registrars
When dialling a number, the nal destination is usually unknown.
There needs to be a network service that tells where a number can be found.
The registrar fulls this role for a specic realm, which is typically bound to a
DNS address.
User agents register with a registrar. When a request for the user
agent arrives at the registrar, it redirects the request to the location that was
previously stored in the internal database.
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Proxies
Proxies forward requests and help the user agent carry out its tasks.
Stateless proxies just forward messages and serve as a “hop” on the path from
a user agent client to a user agent server. The rules for hopping may depend
on all kinds of rules, e.g. traversing NAT using a stateless proxy.
Stateful proxies keep a list of pending requests. This way proxies
can forward requests to different destinations at the same time. When the
responses come back from the destinations, the proxy merges the responses,
determines the best result and passes it down to the user agent that sent the
request (UAC). The snom 4S proxy is a stateful proxy.
Media Server
Strictly speaking, the media server is just a special kind of user
agent. Typically it is able to deal with several calls at the same time and is a
located on a PC or workstation.
The media server has the following tasks:
• Implement mailbox function. When a user is absent, the user agent of
the stateful proxy redirects the call to the mailbox, so that the caller
may leave a message. The owner of the mailbox calls the mailbox
directly to listen to messages.
• Implement music on hold. Using a fat client, all kinds of music tastes
can be played with highest possible quality.
• Implement call parking. Calls can be parked on the media server until
a user picks the call for processing. In the meantime, the caller can
enjoy the music on hold, using DTMF keys to select his favourites.
• Implement conferencing services. Three or more persons dial into
the conference server, which mixes the audio streams for each
participant and also noties them of participants joining and leaving
the conference. The conference server also checks the credentials of
participants joining the conference. The snom 4s is an example of this
technology.
Gateways
From a SIP perspective, the gateway is also just a user agent.
Instead of playing the audio stream on a speaker, it sends it to the PSTN
network and instead of getting voice from a microphone it retrieves signals
from the switched network.
There are three kinds of gateways; PSTN, proxy signalling and NAT

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gateways.
Depending on the nature of the gateway, it may serve one, two, four,
thirty, sixty or more channels at the same time.
Other gateways may translate the signal to existing H323 networks
or other proprietary technology networks. These gateways are sometimes
called signalling gateways. snom does not produce SIP gateways. Examples of
such gateways are manufactured by Cisco, Mediatrix, Sonus and Vegastream.
The snom 4s gateway is a SIP NAT gateway software enabling Linux computers
to be SIP-aware.

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Registrar/Proxy 2.12
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Registrar/Proxy 2.12
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The snom 4S Solution Framework
snom has set up a SIP-based solution targeted at small and medium
sized installations. This solution may be installed on Windows® as well as on
Linux computers. The registar proxy is the core part of this framework.
Proxy
The snom 4S registrar proxy is a SIP registrar and proxy with the
following features:
• Stateful forking. Requests are forked to one or more destinations and
the responses are ltered before passing them back to the user agent
client.
• Sequential forking: Users are searched according to the probability
that was provided with the registration.
• Full functionality: All SIP methods are supported, that includes
transfers, call parking, call picking, notications, instant messaging
and other SIP features.
• Dial plan: You can set up dial plans that will determine whether specic
users may call specic destinations, whether numbers are complete, or
whether numbers are to be redirected to one or more gateways.
• Authentication: You can force clients to authenticate their identity.
• NAT handling: Requests leaving the private network may be redirected
to a NAT gateway.
• Support of path registrations. This way user agents may register with
a path that may contain proxies that must be passed.
• Failure recovery: Even after a reboot, the proxy keeps the state of the
registrations.
• TCP and UDP transport layer support: Both unreliable and reliable
transport layers are supported.3
• Web Access: The proxy can be managed remotely via a web browser.
• Interoperability: The proxy is interoperable with the SIP equipment of
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other vendors; you are not limited to snom products.4
SIP NAT Gateway
The snom 4S Network Address Translation (NAT) gateway is a
stateless proxy that transports SIP messages between private and public
networks. This makes it possible to share one public Internet address amongst
several SIP elements. The NAT gateway supports:
• Forwarding of RTP packets. Both incoming and outgoing packets
may be forwarded by the NAT gateway. The SDP attachments of SIP
messages are patched according to the local ports. This allows usage
of the NAT gateway together with a rewall.
• Path registrations. Registration messages passing the proxy are tagged
with the proxy path.
• Default destination: Packets destined for the NAT gateway may be
forwarded to a xed address. This way a publicly accessible proxy may
reside inside a private network.
• PPPoE device support. In Linux, the NAT gateway automatically detects
the public IP address and changes the address when the PPPoE device
changes the IP address.
• Assignment of RTP port range. To comply with available rewalls, a
range of ports may be assigned.
• Codec preference reordering. The available codecs are reordered
according to their bandwidth requirements. This reduces the bandwidth
used when talking over the NAT gateway and makes usage in DSL
environments easier.
• Linking to Linux ipchains. This way packets destined at SIP port 5060
can be redirected to the NAT gateway without setting up the user
agents in the private network.
Media Server
There are situations when there is nobody available to handle a call.
In these cases the media server helps out.
Other manuals for 4S Media Server
15
Table of contents
Other Snom IP Phone manuals