AltiGen comm MAXCS 82 Softswitch Deployment Guide

							 
MAXCS 8.0 All-Software Solution Deployment Guide  Page 21 of 39 
Outgoing Calls, 3rd-Party IP Phones - Preferred Codec G.729 
First SIP Invite Request Content Codec Used for the Call 
Supports G.729 The IP Codec table of Enterprise Manager is ignored. The system 
uses codec G.729 to negotiate with the endpoint. 
Does not support G.729 No codec is enforced. 
 
Outgoing Calls, 3rd-Party IP Phones - Preferred Codec G.711 Mu-Law 
First SIP Invite Request Content Codec Used for the Call 
Supports G.711 The IP Codec table of Enterprise Manager is ignored. The system 
uses codec G.711 to negotiate with the endpoint. 
Does not support G.711 No codec is enforced. 
 
If the source extension is enabled for Fax-over-IP (FoIP), then no codec is enforced. 
Codec Preference - Outgoing Calls, IP Phones 
Because IP Phone SIP call requests are always from MAXCS, the behavior is different from calls from third-
party phone extensions. 
When a user makes an outgoing call from an IP phone, the system follows the IP Codec table in Enterprise 
Manager. All IP phones support G.729 and G.711 Mu-Law. 
When the Codec Preference feature is enabled, the system uses the following logic for handling outgoing SIP 
trunk calls. 
 
Outgoing Calls, IP Phones 
Preferred Codec Codec Used for the Call 
G.729 
The system modifies the IP phones codec in the RE-INVTE SDP body 
with G.729 codec. After the modification, MaxCS must pick G.729 and 
G.711 codecs as the preferred codecs to negotiate with the endpoint. 
G.711 Mu-Law 
The system modifies the IP phones codec in the RE-INVTE SDP body 
with G.711 codec. After the modification, MaxCS must pick G.711 and 
G.729 codecs as the preferred codecs to negotiate with the endpoint. 
 
If the source extension is enabled for Fax-over-IP (FoIP), then no codec is enforced. 
Step 8: Configure the SIPSP Virtual Board 
A VoIP connection typically consists of two parts. 
Signal Channel – Sets up and tears down a call using protocol. SIP protocol is used in MAXCS to build a 
signal channel between the server and the IP phone.   
						

							 
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Media Path – Encodes, transmits, and decodes voice for both parties. 
The purpose of the virtual board SIPSP is to build signal channels for different connection types, IP extensions, 
SIP Tie Trunks, and SIP Trunking from ITSP. Each channel will have its channel ID similar to channels on a 
Triton extension or trunk board.  
When an IP phone registers to the system, a channel ID will be assigned to the IP extension. However, these 
channels are only responsible for processing protocol and call control signals. They require a media path from 
a VoIP board or from the IP phone to establish a voice steam so that both sides can hear.  
Notes 
 Make sure you have enough VoIP resource boards. 
 The more signal channels in the system, the more system memory and CPU power is required. 
 Changing the number of signal channels requires that you restart the switching and gateway services. 
 SIP Trunking Channel requires a license to activate. 
To configure the SIPSP board, 
1. In MAXCS Administrator, open the Boards panel and double-click SIPSP. 
 
2. Click Board Configuration. The SIP Signaling Channel Configuration panel opens. This panel shows 
the number of configured channels and licensed channels. 
   
						

							 
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3. Adjust the number of SIP Extension channels, Tie-Trunk, and SIP Trunk channels as needed. 
By default, MAXCS is set to support 60 SIP extension channels. The maximum number possible 
depends of the system CPU performance, call volume, and usage.  
If you are using a high performance machine as the Softswitch server, the number of channels can be 
more than 1000.  
Make sure that the system has enough voice processing resources. Calls can fail if there are not 
enough voice processing resources.  
 
4. After you adjust these values, you must restart the switching and gateway services for this change to 
take effect. After the services restart, the new configuration will appear in the Currently Configured 
Channels fields. 
5. After restarting the services, open MAXCS Administrator, double-click SIPSP in the Boards panel and 
then click Advanced Configuration to manage the Trusted SIP Device list. 
To move an IP address from one list to the other, select the IP address and click either the right or left 
arrow button. 
 
Step 9: Block Unauthorized SIP Invite Messages 
You can perform this step now, or return and configure these settings later. 
You can block unauthorized SIP invite messages; this setting is disabled by default. 
When you enable the setting, SIP Invite requests are ignored if the IP address is not configured in one of the 
following places: 
 SIP trunk 
 IP dialing Table 
 Trusted SIP Device List     
						

							 
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IP extensions for the following devices are added to the Trusted SIP Device list automatically, once they 
successfully register to the system (unless they are found in the Malicious SIP Device list: 
 AltiGen IP Phones 
 Third-Party SIP Devices 
 IPTalk 
To enable protection, 
1. In MAXCS Administrator, open Boards view and double-click SIPSP. 
 
2. Click Board Configuration, and then click Advanced Configuration. 
3. Check Block Unauthorized SIP Invite and click OK. 
 
Step 10: Configure the SIP Trunks 
Next, you need to configure SIP trunk channels. Before you start, note the following: 
 If you are using AltiGen SIP Trunks, you may not need a license. For third-party SIP trunk providers, a 
SIP Trunk license is required for each SIP trunk. 
 AltiGen does not guarantee the voice quality of the SIP dial tone coming from your service provider. 
You need to work with your data service and SIP trunking service provider to make sure adequate QoS 
is provisioned for your W AN service. 
 AltiGen does not guarantee SIP trunk implementation will work with all SIP dial tone service providers. 
AltiGen dealers are notified of AltiGen-tested and certified SIP-Trunk service providers. Configuration 
guidelines for each AltiGen-certified SIP-Trunk service provider can be found in the AltiGen authorized 
Partner Knowledge Base, available from the AltiGen Partner Web Site. SIP dial tone service providers 
need to support the following: 
o G.711, G.723.1, G.729 codec 
o RFC 2833 for DTMF tone delivery 
o SIP MD5 authentication with SIP registration 
 If MAXCS is behind NAT, verify that your SIP SP can support this configuration. 
When subscribing to a SIP dial tone service, typically your service provider will provide you with the information 
required in the SIP Trunk Configuration dialog box. Enter the service parameters for each SIP trunk channel 
configuration individually.   
						

							 
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Note:  Make sure that you have enough IP resource boards or that you configure enough HMCP voice 
processing resources to cover your needs. 
To configure a SIP trunk, 
1. In MAXCS Administrator, open the Boards panel and double-click SIPSP.  
2. Click the Board Configuration button. Click the SIP Group Configuration button. 
3. Under SIP Group Configuration, click the Add button to add a SIP group. 
 
4. Enter a name for this SIP group.  
5. If you are using AltiGen SIP Trunks, check the AltiGen Trunk Group option to waive the SIP trunk 
license requirement.  
(If you check this option but are actually using a third-party SIP Trunk, the SIP Trunk will not work.) 
   
						

							 
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6. Highlight the new SIP group (in our example, it is named mysip) and click the lower Add button (the 
one that is below the SIP Servers list) to add a sip server to the group. You can add one or more 
redundant SIP servers, if your service provider has redundant SIP servers. 
 
7. Enter the URL for the domain. Click OK. 
   
						

							 
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8. Highlight the SIP group and then highlight the SIP server that you just added. 
 
9. On the Register tab, enter the parameters as follows. 
 Domain – The Domain Name of the SIP Trunk service provider. For some service provider, it can 
be the same as SIP Server IP Address. 
 SIP Server IP Address – The SIP Trunk service provider’s server IP address. 
 User Name – Assigned by the SIP Trunk service provider. 
 Password – Assigned by the SIP Trunk service provider. 
 SIP Register Period – How frequently the AltiGen system needs to send SIP registration packets to 
the service provider or the SIP gateway. This can detect if the service provider is up or not. Some 
service providers do not accept SIP Register messages. In these cases, you can disable sending 
SIP Register messages from MAXCS by setting the SIP Register Period to 0. 
 SIP Source Port – For SIP UDP, select the source port from 5060 or 10060. For TCP or TLS, you 
cannot change ports. Using a port other than 5060 will prevent SIP-ALG firewall/router from 
changing the SIP packets. Port 10060 is recommended, if the service provider supports it. 
 SIP Destination Port – A SIP Trunk can have different source port and destination port.   
						

							 
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10. Click Channel Assignment to assign channels to the SIP Group that you just configured. 
 
11. Highlight the trunk channels and press Assign Group to assign them to the SIP Trunk group. License 
usage will be automatically updated. SIP trunk service provided by AltiGen does not require license 
when AltiGen SIP trunk group is checked. 
 
12. Highlight these SIP trunk channels and check their boxes to enable them.   
						

							 
MAXCS 8.0 All-Software Solution Deployment Guide  Page 29 of 39 
 
13. In Enterprise Manager, add the SIP Trunk service provider’s IP address to the IP Device Range (on 
the Servers > IP codec tab) and select the proper codec profile for this service.  
If you omit this step, then calls may have no audio even if the SIP Trunk channel shows the call is 
connected.  
Step 11: Enable SIP Option (optional) 
SIP Option is a keepalive message to check the SIP server’s availability.  
If the server or network between the SIP server and MaxCS is down. MaxCS will mark the server as down if 
either SIP registration or SIP Option fails. If all SIP servers with in the same SIP group are down, the all the 
trunks associated with the SIP group will become not ready and the status in the list will show as DOW N. 
 
1. Highlight the SIP Group and then highlight the SIP Server.  
2. Switch to the SIP OPTIONS tab. 
 
You can enable or disable SIP OPTIONS independently for each SIP Server. By default, the SIP 
Options feature is disabled.   
						

							 
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3. Configure the parameters as follows, and click OK. 
 SIP Server IP Address – The IP address for this SIP Server. 
 Enable SIP OPTIONS – Enable or disable this feature for the selected SIP Trunk group. 
 SIP Options Interval – How often, in seconds, the server sends a “keepalive” message to this 
enabled SIP trunk group; the default interval is 30 seconds. 
 Number of Retries – If MaxCS receives no 200 (OK) response, the number of times a “keepalive’ 
message should be sent. After these retries, if there still has been no valid response, then MaxCS 
marks all SIP trunks in the group as Not Ready. The default number of attempts is 5. 
 Retry Interval – While a SIP trunk group is in a Retry state and is not receiving a valid response, 
how often MaxCS should send another “keepalive” message to the SIP server. The default interval 
is 2 seconds. 
 If MaxCS does not receive a SIP 200 (OK) message after the set number of retries, it then sets all 
SIP trunks in that group to Not Ready. 
Step 12: SIP Trunk Setting 
Different SIP service providers may have different settings such as supporting different ways of sending a 
caller ID and so on. These options are found in the SIP server’s Settings tab.  
1. Highlight the SIP group and then highlight the SIP server in that group (this allows you to configure the 
server). 
 
2. Edit the fields as necessary and click OK. (Refer to the following table for descriptions of these fields).