SIP2SIP

h1. Acrobits Groundwire

h2. Account

h3. User Details

|Username| 2233XXXXXX|

|Domain|sip2sip.info|

|Display Name|<empty>|

h3. 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._

h4. Backgrounding Options

|Host|<empty>|

|Transport Protocol|tls (sips)|

|Expires|86400|

|Track Errors|On|

h4. 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|

h4. Codecs for WiFi

# iLBC

# G.711 u-Law

# G.711 a-Law

# G.722

# Opus Wideband

# Opus Narrowband

# G.729a

# 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._

h4. Codecs for 3G

# iLBC

# Opus Narrowband

# GSM

# G.729a

# G.711 u-Law

# G.711 a-Law

# Opus Wideband

# G.722

|Packet Time|20ms|

|Force Packet Time|Off|

|Honor Remote Codecs|Off|

h4. DTMF Mode

h5. Enabled DTMF Modes

# RFC2833

# SIP INFO

# 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._

h4. Secure Calls

h5. 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._

h5. 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._

h4. Hacks

|RTP Port Start|10000|

|RTP Port End|65535|

|SIP Port|<empty>|

h5. Forced Contact

|Contact IP:port|<empty>|

h5. Authorization

|Send On Request|Off|

|URI Scheme|sip:|

h5. Nortelnetworks

|Proxy-Require|Off|

h5. SRTP

|Prefer 80-bit Tags|Off|

_Prefer 80-bit authentication tags over 32-bit tags._

h5. 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._

h5. Well-known codecs

|Use rtpmap|Off|

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