baresip Baresip based Android application\n\n Work in progress\n Juha Heinanen jh@tutpro.com\n\n Usage Hints\n\n - start by creating an account (click item titles for help)\n - usually there is no need to edit the config\n - history entries can be removed or added to contacts by long clicks\n - call click when callee has not been given can be used for re-dial\n\n No accounts Name (if any) used in From URI of outbound requests. Authentication username if required by outbound proxy. Authentication password if required by outbound proxy. SIP URI of one or two proxies that must be used when sending requests. If two is given, REGISTER requests are sent to both and other requests are sent to one that responds. If no outbound proxy is given, requests are sent based on DNS NAPTR/SRV/A record lookup of callee URI hostpart.\nExample:\nsip:foo.com:5060;transport=tls Tells how often (in seconds) REGISTER requests are sent. Value 0 disables registration. List of supported audio codecs in priority order. Selects media transport encryption protocol.\n ZRTP (recommended) means that ZRTP end-to-end media encryption negotiation is tried after the call has been established.\n DTLS-SRTPF means that UDP/TLS/RTP/SAVPF is offered in outgoing call and that RTP/SAVP, RTP/SAVPF, UDP/TLS/RTP/SAVP, or UDP/TLS/RTP/SAVPF is used if offered in incoming call.\n SRTP-MAND means that RTP/SAVP is offered in outgoing call and that RTP/SAVP or RTP/SAVPF is used if offered in incoming call.\n SRTP means that RTP/AVP is offered in outgoing call and that RTP/SAVP or RTP/SAVPF is used if offered in incoming call.\n EMPTY value means that media encryption is not supported and that media transport protocol thus is RTP/AVP.