Goals of the demo :
Use asterisk like a UPD/TCP translator between OCS and a SIP trunking services in UDP mode.
I use Asterisk 1.6 which support TCP and UDP installed on CentOS 5 – Kernel 2.6.18
Installation steps :
OCS R2 —- Mediation —- Asterisk —— Firewall — SIPTrunk
MTLS —- TCP —- UDP —- NAT —– UDP
I use Hyper-V for the two OCS servers and VMware for the Asterisk server.
The first will be use for internal numbering and the second is a generic rule that redirect all call that do not correspond to a valid extension number in OCS to the default gateway.
Phone pattern : ^2(\d{2})$
Translation pattern : +2$1
Phone pattern : ^(.*)$
Translation pattern : $1
Assign the location profile to the front-end server (properties of the pool / properties of front-end / Voice Tab)
NB : Test your dial plan using the Enterprise Voice route helper
tcpenable = yes
bindport = 5060
nat=yes
externip=Valid_FQDN
localnet=Your_Localnet/Your_Subnet
Trunk Name : ocs
PEER Details :
host=ip-mediation-server
type=peer
qualify=yes
transport=tcp
insecure=very
port=5060
canreinvite=yes
fromdomain=yourdomain
context=from-ocs
Incoming Settings :
User context : (I put a OCS username here)
User details :
host=ip-mediation-server
type=peer
transport=tcp
insecure=very
port=5060
context=from-ocs
register string :
(leave blank)
Trunk Name : siptrunk
PEER Details :
type=friend
disallow=all
allow=ilbc&speex&gsm&alaw&ulaw
username=username
secret=password
host=yoursipregistrar
canreinvite=no
context=from-siptrunk
Incoming Settings :
Clear all
register string :
username:password@yoursipregistrar (/yournumber if needded)
Edit Extension_Custom.conf and add a the end of the file :
[from-ocs]
exten => _X.,1,Answer
exten => _X.,2,Dial(SIP/${EXTEN}@siptrunk,,tr)
#include extensions-away-status.conf
[from-siptrunk]
exten => _X.,1,Set(numDialled=+${EXTEN:Number_of_X_to_ignore})
exten => _X.,2,Set(__FROM_DID=${EXTEN})
exten => _X.,3,Answer
exten => _X.,4,Dial(SIP/${numDialled}@ocs,,tr)
exten => _X.,4,Dial(SIP/${numDialled}@ocs)
Assign line number from your sip trunk to Asterisk extension and redirect to phone extension in OCS by using a # after the number.
DID number from your SIP trunk provider —> Extension in Asterisk —> Follow-me to OCS extention (use # after the number to precise that it’s external to asterisk)
Have fun with that and leave me a message if encounter some problems!
It’s probably possible to do the same with other IP/PBX like : Freeswitch, OpenSer, SipxECS, …
Date : 04/08:2009 - Author : Yann Espanet – mail : yann@unifiedcommunications.eu
Trouble to use regular expressions in defining routes for OCS ? Download a whitepaper from Globalknowledge and a cheatsheet from addedbytes !
“Microsoft has introduced regular expressions for the main purpose of normalizing E.164 numbers and allowing users to dial numbers by a pattern they are accustomed to, and to define routes to send to an external gateway for PSTN connectivity. This white paper focuses on the regular expression process and the syntax used by the Microsoft OCS (Optional Component Manager) Expert to create a dial plan and normalization rules that will properly be interpreted and executed. Regular expressions are also used for Address Book translations of numbers in the user contact database that would have to be converted to the E.164 format. This paper also examines tool sets that can be used right on your XP or Vista computer to test regular expression constructs without disturbing the corporate production environment.”
http://images.globalknowledge.com/wwwimages/whitepaperpdf/WP_Parlas_MicrosoftOCS.pdf
http://www.addedbytes.com/download/regular-expressions-cheat-sheet-v2/pdf/
The maximum message size for a UDP datagram socket is limited by the maximum size of an IP datagram and the size of the UDP datagram socket buffer.
The maximum size of an IP datagram limits the maximum message size of a UDP message to 65507 bytes. Therefore, using the maximum socket buffer size will allow multiple maximum-sized messages to be placed on the send queue. The default inbound and outbound message size limit for a UDP datagram socket is 65535 bytes.
The maximum message size for a UDP broadcast is limited by the MTU size of the underlying link, which is 1500 on Ehternet.
Here’s the MTU for different media :
Network MTU
Token Ring 16 Mb/s 17 914
FDDI 4 352
Ethernet 1 500
IEEE 802.3/802.2 1 492
PPPoE 1 480
X.25 576
UDP has no fragmentation mechanism like TCP or IP, but it used the fragmentation of IP layer.
Transmission Control Protocol (TCP) is an example of a protocol that will adjust its segment size to be smaller than the MTU.
User Datagram Protocol (UDP) disregard MTU size thereby forcing IP to fragment oversized datagrams.
Conclusion :
TCP provide transport-layer fragmentation. If a SIP message is larger than the MTU size, it is fragmented at the transport layer. When UDP is used, fragmentation occurs at the IP layer. IP fragmentation increases the likelihood of having packet losses and makes NAT and firewall traversal difficult, if not impossible. This feature will become important if the size of SIP messages grows dramatically.
While the fragments above are in transit, they may need to pass over a hop between two routers where the physical network’s MTU is only 1,300 bytes. In this case, each of the fragments will again need to be fragmented. The 3,300 byte fragments will end up in three pieces each (two of about 1,300 bytes and one of around 700 bytes) and the final 2,100-byte fragment will become a 1300-byte and 800-byte fragment. So instead of having four fragments, we will end up with eleven.
This example shows illustrates a two-step fragmentation of a large IP datagram. The boxes represent datagrams or datagram fragments and are shown to scale. The original datagram is 12,000 bytes in size, represented by the large gray box. To transmit this data over the first local link, Device A splits it into four fragments, shown at left in four primary colors. The first router must fragment each of these into smaller fragments to send them over the 1,300-byte MTU link, as shown on the bottom. Note that the second router does not reassemble the 1,300-byte fragments, even though its link to Device B has an MTU of 3,300 bytes.
Today’s SIP application is mostly operating over the unreliable transport protocol UDP. In lossy environment such as wireless networks and congested Internet networks, SIP messages can be lost or delivered out of sequence. The SIP application then has to retransmit the lost messages and re-order the received packets. SIP has a builtin process for dealing with the unreliable nature of UDP, however to parse the SIP data a transition from kernel to usermode is required. This additional processing overhead can degrade the performance of the SIP application.
Conclusion :
It’s not a network problem, but a CPU problem !
Take a look at this study :
http://www.research.ibm.com/people/n/nahum/papers/sigmetrics07-sip-perf.pdf
Ref : RFC 4168
http://www.rfc-editor.org/rfc/rfc4168.txt
Therefore to solve this problem, the researchers are looking for a more appropriate transport layer for SIP. SCTP, a transport protocol providing acknowledged, error-free, non-duplicated transfer of messages, has been proposed to be an alternative to UDP and TCP. The multi-streaming and multi-homing features of SCTP are especially attractive for applications that have stringent performance and high reliability requirements and an example is the SIP proxy server.
But additional testing need to be done, and Microsoft doesn’t support SCTP for the moment !
Another article can be found there :
Increased SIP Performance with Stream Control Transmission Protocol
http://blog.tmcnet.com/cross-talk/2008/07/increased-sip-performance-with-stream-control-transmission-protocol.html
SCTP and its applications
http://www.apng.org/9thcamp/Slide/Satit.pdf
Right in OCS but not in his small business product !, Microsoft SIP Solution for Small Business support UDP !!
http://www.microsoft.com.nsatc.net/responsepoint/resources-whitepapers-sbsintegration.aspx
“Microsoft Response Point is complete phone system software designed specifically for small businesses with 1-50 employees. In the past, deploying an advanced phone system to a small business was considered impractical.”
“Network Protocols
Response Point transfers data and communicates using standard protocols. Generally, there should not be any network configuration changes needed to support these protocols. All hardware and software connected to your network—the router, SBS, and other computers— should work automatically with the following protocols:
Session Initiation Protocol (SIP over UDP) to establish and coordinate phone calls between devices.
Screenshot of trying to install OCS R2 64bit 3.5.6907.0 on Windows 2008 R2 Standard 6.1.7100 Build 7100 …..
Adam Glick from Microsoft demonstrates the newest Exchange 2010 features for Windows Mobile 6.1/6.5 :
Exchange 2010 Unified Messaging is Microsoft’s second generation unified messaging and voice mail solution. In this webcast, learn about the features, …
Exchange 2010 carries on the tradition that began in Exchange Server 2007 of dramatically reducing mailbox storage costs. Attend this webcast and learn how …
In this webcast, we reveal the changes and improvements to the built-in high availability platform in Exchange Server 2010. Exchange 2010 includes a unified …
19 Jun 2009 … The upcoming release of Exchange 2010 can help you achieve better business outcomes while controlling the costs of deployment, …
Exchange 2010 includes new capabilities that make the operation of your Exchange environment more efficient. Learn how we’ve made the Exchange Management …
This event will provide you with important information on the changes made to the upcoming Exchange Server 2010. The discussion will focus on how Microsoft …
17 Jun 2009 … They wish to install Exchange Server 2010 on a single server. Northwind Traders must make sure that the server is adequately prepared and …
Microsoft’s product plans for the next release code named “Exchange 2010 focuses around three … o Operational Flexibility – With Exchange Server 2010, …
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Microsoft® My Phone syncs information between your mobile phone and the web, enabling you to:
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Microsoft has announced that they upgraded the My Phone website to support 19 additional languages plus some improvements based on users feedback. There is no details about the improvements. From now on, the My Phone beta is open to the public.
Promotional codes are no longer needed !
To start using the service, point your Windows Phone’s browser to http://myphone.microsoft.com.
Communicator KB Exchange 2007 KB Communications Server KB
