Some advanced configuration settings about the PrivateServer behavior.
2.4.1 SIP/TLS
SIP/TLS is about configuring the encrypted communication channel among PrivateServer and its clients. The configuration form is reachable by the SIP/TLS main menu entry.
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From the
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2.4.2 RTP
"The Real-time Transport Protocol (RTP) defines a standardized packet format for delivering audio and video over IP networks" (quote from Wikipedia).
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In this form that you get by the RTP main menu entry, you can set up the voice transport features. Rtpstart and Rtpend are define the number of RTP ports available for the calls. RTP port range reserved for RTP traffic.
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Each call uses 4 UDP ports are reserved for each concurrent call, thus you can do your math on the RTP number necessary range extension required in your configuration, multiplying the number of foreseen concurrent calls for by 4. |
In the example shown in
you see:Xref AnchorName rtp form
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Strictrtp Enables the strict RTP protection. This will drop RTP packets that do not come from the source of the RTP stream. This option is disabled by default.
2.4.3 Jitter Buffer
Jitter is the undesired deviation from true periodicity of an assumed periodic voice streaming. The variation in latency as measured in the variability over time of the packet latency across a network. The consequences of jitter, often called jittering, are a voice communication with holes gaps in it or stirring metal voice effect. Mostly on a 3/4G GPRS and EDGE network (and in general in a mobile network environment), the jitter is a sensible problem to face. To avoid jitter issues a jitter buffer is implemented in PrivateServer. cope with jittering issues, a jitter buffer produces a smooth and regular audio output, just adding some more latency.
PrivateGSM and VoIP devices already have an embedded jitter buffer, so it is not required to enable it on PrivateServer also. For very old devices and SIP trunks which have an inefficient jitter buffer, it is possible to enable it on PrivateServer.
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To apply your changes just press the Update button and the management interface will ask you to restart the asterisk service in order to apply your new configuration, as shown in
2.4.4 Obfuscation
The Obfuscation is an internal VoIP communication stealth mode. It This is a useful to avoid QoS (Quality of Service) checks on VoIP countermeasure to bypass VoIP blocks and censorship, as it masks the data.
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| This practice is legal if you are not fooling your mobile provider or cheating your network administrator. |
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To avoid calls problems such as abruptly interrupted calls you make sure the obfuscation mode and key are equally set up on the server and the clients. |
2.4.5 NAT Configuration
If you are using the appliance in an internal network then it's most possible that you need to configure the NAT option. NAT stands for Network Address Translation and it's commonly used to let services on a private IP address to be reachable by a public IP address.
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Apart from your router/firewall configuration (please check PrivateGSM installation pre-requisites) and your network design/topology, from the PrivateServer point of view the only known thing is that the appliance is configured on a private IP address but the requests of the encrypted voice service are made to an external and public IP address. To avoid wrong replies the PrivateServer must know of this setup and be configured accordingly. Thus if you fall in the described scenario access to the "NAT Configuration" form (showed in
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By default this option is disabled, so to enable it you first need to select "YES" in "NAT" option. If you have enabled the NAT then it's mandatory to configure the remaining options as well.
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The "Keep-alive Frequency" is part of a new feature that is not directly connected to the NAT setup. To better understand what a keep-alive is, please refer to PSOM 1.0 Groups. |
External media address
This is the public IP address used for the RTP delivery. It means that this is the secured voice IP you want to use.
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| Unless you need to specify for some reason a specific IP address for RTP, you'd better leave this field empty and let Asterisk do the job for you! |
External SIP address
This is the public IP address used for the SIP delivery. It means that this is the IP you want to use for SIP signalling.
External port
If you want to perform a PAT (Port Address Translation) in addition to the NAT, then please use this option to explain to the appliance which port number is used on the external interface for providing the encrypted SIP service.
2.4.5.1 Keep-alive Frequency
If you are using the keep-alive option (please refer to PSOM 1.0 Groups) then you may find this option handy. You can define here how many seconds should pass between each keep-alive request sent by the server to each client configured with the keep-alive option.
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If any mobile user has been configured with the keep-alive option on, then we strongly suggest you to set the keep-alive Frequency to 180 seconds (i.e. 3 minutes) at least in order to save battery life.
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