Acrobits Groundwire¶

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
STUN Server	stun2.dns-hosting.info:3478
TURN Username	<empty>
TURN Password	<empty>
Ignore Symmetric NAT	Off
Discover Global IP	Internal
Send keepalives	Off
Outbound Proxy Enabled	Off

Codecs for WiFi¶

1. iLBC
2. G.711 u-Law
3. G.711 a-Law
4. G.722
5. Opus Wideband
6. Opus Narrowband
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.