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Some advanced configuration settings about the

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behavior.

2.4.1 SIP/TLS

SIP/TLS is about configuring the encrypted communication channel among

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and its clients. The configuration form is reachable by the SIP/TLS main menu entry.

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Prefix figure SubtitleText Cipher list configuration form AnchorName siptls_conf
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From the

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AnchorName siptls_conf

you can set up the  cypher list of the

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. This is the list of accepted cipher suite, using OpenSSL format. Check at http://www.openssl.org/docs/apps/ciphers.html#CIPHER_LIST_FORMAT. Usually you can leave the default values.

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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Prefix figure SubtitleText RTP configuration form AnchorName rtp form

In this above form that (the one you get by the clicking on RTP main menu entry), you can set up the voice transport features. 

Rtpstart and RtpEnd

Rtpstart and Rtpend define the RTP port range reserved for RTP traffic.

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18000 - 15000 = 3000 ports available. This means 3000/4 = 750 concurrent calls threshold. 

Rtpchecksums

The Rtpchecksums enable the application checksum over UDP encrypted voice transmission. This is an error detection commodity, which adds 16 bits per packets payload.

Strictrtp

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. 

Note
In order to enable call roaming through different kind of networks (mobiles 'X'G family and Wi-Fi) you need to DISABLE Strictrtp.

Strictrtp option is about call roaming. If it's enabled then each call would be terminated on network change event.

Rtp timeout

Rtp timeout sets up the amount of seconds one server can wait without dealing RTP packages on one call's leg. This value is connected to client's setup and it gives you the fault tolerance time of one call. 

Info
Rtp timeout is used in combination with Strictrtp to enable call roaming: during network change events RTP stream is lost until one new UDP connection is established between client and server. Rtp timeout is the maximum amount of seconds one server can wait before considering such connection as lost and thus hang up any related ongoing call.

2.4.3 Jitter Buffer

Jitter is 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 gaps in it or stirring metal voice effect. Mostly on a GPRS and EDGE network (and in general in a mobile network environment), the jitter is a sensible problem to face. To cope with jittering issues,  a jitter buffer produces a smooth and regular audio output, just adding some more latency.

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Prefix figure SubtitleText Jitter Buffer configuration form AnchorName edit jitter buffer
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The jitter buffer is an elastic buffer in which the signal is temporarily stored and then retransmitted at a rate based on the average rate of the incoming signal. Checking the Enable checkbox you allow

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to perform such jitter buffering in general, when needed.

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Prefix figure SubtitleText The Management console asks to restart the service AnchorName restart service
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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. This is a useful countermeasure to bypass VoIP blocks and censorship, as it masks the data.

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Prefix figure SubtitleText Edit the obfuscation parameters AnchorName edit_obfuscation
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The configuration is quite simple. Mode enables/disables the obfuscation mode.

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Tip
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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Prefix figure SubtitleText NAT configuration form AnchorName nat_config
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Apart from your router/firewall configuration (please check <ac:macro ac:name=) and configuration and your network design/topology, from

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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

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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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AnchorName nat_config

) using the "NAT" link under "Server Configuration".

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Info
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.

Warning
title Possible Misconfiguration
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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