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Pro Backup, 08/05/2013 08:18 pm
shared with Snom added

SipDevices »

Acrobits Groundwire

Goal is to create a reliable and secure configuration for inbound calls, where the account is shared with another desktop (Snom) SIP-phone.

Account

User Details

Username 2233XXXXXX
Domain sip2sip.info
Display Name <empty>

Advanced settings

Incoming Calls On with Backgrounding
NAT Traversal ICE
Proxy proxy.sipthor.net:443
Auth User Name <empty>
Transport Protocol tls (sips)
VoiceMail Number <empty>
Expires 86400
Caller ID <empty>
Caller Id Method From Username

Sets caller ID headers of outgoing INVITE messages. Most VoIP providers will ignore these headers though.

SIMPLE Off

Enables SIP/SIMPLE messages to be sent and received.

Backgrounding Options

Host <empty>
Transport Protocol tls (sips)
Expires 86400
Track Errors On

NAT Traversal

Media ICE
Send Media Back Off

Ensures that media streams are sent to the IP:port they are received from.

STUN Server stun1.dns-hosting.info:3478
TURN Username <empty>
TURN Password <empty>
Ignore Symmetric NAT Off
Discover Global IP Internal

For providers requiring global IP discovery use External. For others you should be safe with Static or Internal. Static uses a faked private IP:port pair to solve issues when you are behind NAT and your provider groups registrations based on the full Contact URI. This option prevents changes in Contact header when network conditions change. To specify your own IP:port pair go to the Hacks section.

Send keepalives Off

Send keepalive packets to keep the NAT ports open. Set if you have troubles getting incoming calls.

Outbound Proxy Enabled Off

Optionally set to a proxy that performs some traffic manipulation e.g. TLS to UDP translation. This proxy is usually not related to the SIP provider. If regular proxy is specified as well, the SIP traffic will be routed to the proxy on the second hop.
Use the Outbound Proxy to route all SIP traffic through one server.

Codecs for WiFi

  1. Opus Wideband
  2. G.722
  3. Opus Narrowband
  4. iLBC
  5. G.711 u-Law
  6. G.711 a-Law
  7. G.729a
  8. GSM
Packet Time 20ms
Force Packet Time Off

Higher packet times save bandwidth, but make the call quality more sensitive to packet loss. Bigger lost packets will make longer, more audible dropouts.

Honor Remote Codecs On

If set, local codec order will be ignored and the first supported codec from the list sent by remote will be used.

Codecs for 3G

  1. iLBC
  2. Opus Narrowband
  3. GSM
  4. G.729a
  5. G.711 u-Law
  6. G.711 a-Law
  7. Opus Wideband
  8. G.722
Packet Time 20ms
Force Packet Time Off
Honor Remote Codecs Off

DTMF Mode

Enabled DTMF Modes
  1. RFC2833
  2. SIP INFO
  3. audio
Send All Enabled Off

When turned on all enabled DTMF methods are sent simultaneously. If pressing a single digit results in multiple presses on the receiver side, just turn this switch off.

Secure Calls

SDES (RFC 4568)
Incoming Calls Enabled
Outgoing Calls Best Effort

A secure signaling channel is required for SDES. I.e. the protocol needs to be set to TLS. Due to security concerns it's also disabled for pushed calls. Please note SDES is prone to man-in-the-middle attacks and is largely dependant on the behavior of proxies along the SIP path. There may be hops between which the keys are transfered in clear text. If you would like to use SRTP over an insecure signaling channel you should try ZRTP instead.

ZRTP
Incoming Calls Enable
Outgoing Calls Disable

ZRTP is a media path key exchange method for SRTP. It can be used to secure calls even over insecure signaling channel (e.g. UDP). As opposed to SDES, it prevents eavesdropping opportunities at proxies. You will be able to accept ZRTP encrypted calls, however to initiate them you need to purchase the ZRTP add-on.

Hacks

RTP Port Start 10000
RTP Port End 65535
SIP Port <empty>

The listening port for SIP.

Forced Contact
Contact IP:port <empty>

Fake fixed local IP and port can help if you are behind NAT and your provider groups registrations based on the full Contact URI. This setting prevents changes in Contact header when network conditions change. You should pick an IP from RFC 1918 range. E.g. 192.168.1.100:44444. Make sure to set Discover Global IP to Static to use this field.

Authorization
Send On Request Off

Some providers do not like getting the Authorization header unless requested.

URI Scheme sip:

The default scheme for numerical URIs is sip: or sips:. You can select tel: to enable support for RFC 3966 tel: URIs.

Nortelnetworks
Proxy-Require Off

Some setups need Proxy-Require: com.nortelnetworks.firewall header to successfully traverse NAT.

SRTP
Prefer 80-bit Tags Off

Prefer 80-bit authentication tags over 32-bit tags.

Registration State
Reuse On
Adjust Via Off

The new registration will reuse the same Call-ID, CSeq sequence and rinstance as the previous one. We try to unregister stale contacts when network change. It's possible to alter Via headers to reflect Contact being unregistered.

Well-known codecs
Use rtpmap Off

It's not necessary to include rtpmap attributes for well known codecs, but some providers erroneously require it.