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31 Mar

By default no configuration file is required and linphone always generates a default one at first startup.

Most of the parameters are used internally by accessors of liblinphone (ex: linphone_core_set/get_audio_port()) and thus do not need to be accessed directly by editing the config file.

For those who are behind a proxy: You can change the access method from git:// to as follows: Normal git access:git clone git://git.linphone.org/Behind a proxy, use http access:git clone day some binary packages generated from the current git state are uploaded at .

It is worth to visit it if you want to test and see the latest changes, especially when a long time has elapsed since the last official stable release.

These snapshots are generated automatically and not tested, some features might be broken from time to time.

help Print commands helpcall Call a SIP urichat Chat with a SIP uriterminate Terminate the current callanswer Answer a callautoanswer Show/set auto-answer modeproxy Manage proxiessoundcard Manage soundcardswebcam Manage webcamsstaticpic Manage static pictures when nowebcamipv6 Use IPV6refer Refer the current call to the specified Set nat addressstun Set stun server addressfirewall Set firewall policycall-logs Calls historyfriend Manage friendsplay play from a wav filerecord record to a wav filequit Exit linphonecregister Register in one line to a proxyunregister Unregister from default proxyduration Print duration in seconds of the last call.status Print various status informationports Network ports configurationspeak Speak a sentence using espeak TTS enginecodec Audio codec configurationvcodec Video codec configurationec Echo cancellationmute Mute microphone and suspend voice transmission.unmute Unmute microphone and resume voice transmission.nortp-on-a Set the rtp_no_xmit_on_audio_mute configuration parameter Type 'help #spawn a linphonec daemon:$ linphonecsh init#register to a proxy$ linphonecsh register --host --username bill --password thisisasecret#dial out$ linphonecsh dial "sip:[email protected]"#terminate the call$ linphonecsh hangup#linphonecsh can transmit any command understood by linphonec using the 'generic' keyword$ linphonecsh generic "proxy list"#exit the daemon$ linphonecsh exit Linphone has a linphonerc configuration file in which all SIP, RTP, sound, video settings are read and stored.

[sound]#The noise gate is located just after mic input#Tells whether noise gate is active:noisegate=0#Noise gate threshold in linear power between 0 and 1:#Above this threshold the noise gate becomes bypass.ng_thres=0.05#Noise gate's floorgain: gain applied to the signal when its energy is below the threshold.#It is expect to be low so that noise is attenuated.ng_floorgain=0.0005[sound]#Automatic gain control (of mic input) - turns on or off:agc=0#static software gain (linear scale) to be applied to microphone signalmic_gain=1.0#static software gain (log scale) to be applied to signal sent to speakerplayback_gain_db=0.0#Enable or disable DC removal of mic input:dc_removal=0##section relative to sound settings[sound]#sound device name used for playback, as listed in linphonec by "soundcard list"playback_dev_id=ALSA: default device#sound device name used for ringing, as listed in linphonec by "soundcard list"ringer_dev_id=ALSA: default device#sound device name used for capture, as listed in linphonec by "soundcard list"capture_dev_id=ALSA: default device#Alsa special device name# This option allows to specify a special ALSA card (as defined in ALSA asoundrc config files)# to be used by linphone.

This card can then be referred by playback_dev_id, ringer_dev_id, capture_dev_id options.# Use this if you are able to understand asoundrc syntax and you know what you are doing.#alsadev=#wav file to play to advertise remote ringingremote_ring=/usr/local/share/sounds/linphone/ringback.wav#wav file to play to advertise incoming callslocal_ring=/usr/loca/share/sounds/linphone/rings/bigben.wav#turn on/off echo cancellationechocancellation=1#Expected delay of echo in milliseconds#Use this when you have a fixed latency in the sound hardware.#This allows to reduce the tail length of the echo canceller, which speeds up convergence#and reduces complexity of computations.ec_delay=0#Tail length of echo canceller in milliseconds.#Ideally it should be no more than the expected duration of the echo.ec_tail_len=60#Frame size for AU-MDF echo canceller algorithm#This is a parameter internal to the echo canceller, recommended is too keep to its default value.ec_frame_size=128#static software gain (linear scale) to be applied to microphone signalmic_gain=1.0#static software gain (log scale) to be applied to signal sent to speakerplayback_gain_db=0.0##Video settings[video]#Size of sent video among these names: QCIF, QVGA, CIF, VGA, SVGAsize=cif#Whether video is enabled:enabled=1#You can refine whether it is enabled for display or capture or bothdisplay=1capture=1#Show local preview between calls.show_local=1#Show local view during calls, in a corner of the video windowself_view=1#Webcam name for capturedevice=V4L2: /dev/video0##Network settings[net]#Estimated download bandwidth in kbit/sdownload_bw=1024#The bandwidth settings are used to control the bitrate of video (and sometimes audio) encoder, as well#as limiting the size or fps of the sent video.#Estimated upload bandwidth in kbit/supload_bw=1024#Firewall policy:# 0: assume there is no nat# 1: use firewall address supplied in "nat_address" item (discouraged)# 2: use STUN to discover its own public IP address and ports# 3: use ICE.firewall_policy=0#Network's Maximum Transmission Unit# Use 0 to allow automatic discovery, otherwise set to a number of bytes.# This parameter is only meaningful with video streams for which RTP packets are big.mtu=0#STUN server address to use when in firewall_policy=2stun_server=stun.ekiga.net#Firewall address to use when in firewall_policy=1nat_address=80.1##SIP settings[sip]#SIP port usedsip_port=5060#Discover automatically local IP addressguess_hostname=1#Contact address when no proxy is used# The host port is always overriden at runtime if guess_hostname # is set to 1.contact="Bob" #Incoming call answering timeoutinc_timeout=15#Use SIP INFO to send DTMFs (digits)use_info=0#Use RFC2833 (out of band DTMFs) to send digitsuse_rfc2833=0#Use IPv6. use_ipv6=0#Send registers only when network is upregister_only_when_network_is_up=1#Default proxy to use (the number is the index of the proxy configuration in this config file)# Use -1 for no proxy.default_proxy=-1#Keepalive period in milliseconds for sending out SIP UDP keepalive to the proxies.keepalive_period=10000#When answering to SDP offers, select only one codec, #instead of replying with all matching codecs.only_one_codec=0#Send an OPTIONS message before doing outgoing calls#This is used by Linphone to workaround some NAT problems inherent to SIP.#This is highly recommended.ping_with_options=1#Network state automatic monitoring# When set to 1, linphone will periodically monitor the network state (by checking whether it is possible# to reach the internet).# When the operating system has callbacks to notify such information, you can use # linphone_core_set_network_reachable() to notify the core, in which case no network monitoring will be done internally.auto_net_state_mon=1## RTP settings[rtp]#Audio RTP (UDP) portaudio_rtp_port=7078#Video RTP (UDP) portvideo_rtp_port=9078#Nominal audio jitter buffer size in millisecondsaudio_jitt_comp=60#Nominal video jitter buffer size in millisecondsvideo_jitt_comp=60#RTP timeout in seconds: when no RTP or RTCP# packets are received for this period, the running call is # automatically closed.nortp_timeout=30## Audio codec descriptions# These sections are named audio_codec_X, where X is a number.# This number identifies the position of the described codec# in the core's audio codec list.[audio_codec_0]# sub-mime type as defined in RFC3551 or codec's specific RFC:mime=speex# RTP clock-rate as defined in RFC3551 or codec's specific RFC:rate=8000# Tells whether is codec is enabledenabled=1# Fmtp (format parameters) string to be sent in SDP for this codec, which# corresponds usually to what we are prefering to receive.# RFC3551 or codec's specific RFC describes the allowed parameters.recv_fmtp=vad=on## Video codec descriptions# These sections are named video_codec_X, where X is a number.# This number identifies the position of the described codec# in the core's video codec list.[video_codec_0]# sub-mime type as defined in RFC3551 or codec's specific RFC:mime=H264# RTP clockrate as defined in RFC3551 or codec's specific RFC, usually 90000 for video payloads.rate=90000# Tells whether the codec is enabledenabled=1# Fmtp (format paramters) string to be sent in SDP for this codec, which# corresponds usually to what we are prefering to receive.recv_fmtp=packetization-mode=1## SIP Proxy configuration# Like with audio_codec, it is possible to define several proxy configuration in the# form of [proxy_X] section, where X is a number.[proxy_0]#SIP address of the proxyreg_proxy=sip:example.net#SIP identity for which you are known on this proxy:reg_identity=#Expiration period of the registration in secondsreg_expires=3600#Whether to send a register or notreg_sendregister=0#Route: SIP server address to send all outgoing SIP requests#It is usually left blank, otherwise it is commonly used to specify this proxy #must be used as an outbound proxy, for example:# reg_route=sip:reg_route=#Send a PUBLISH request to the proxy to notify about presence information (online, busy, out to lunch)publish=0#whether " " in phone numbers should be replaced by 00dial_escape_plus=0#Phone number prefix to be applied to entered destinations.#Example: prefix= 33prefix=## Authentication information# Similarly, several auth_info_X can be defined# Authentication information is kept distinct from proxy information# because there can be authentication challenges from proxies or user# agents even if we are not registered to any proxy.[auth_info_0]#SIP usernameusername=bob#sip userid (usually the same as username, don't specify unless you know what you are doing)userid=bob#password associated with above username, userid and realmpasswd=mysecret#SIP authentication realm (= authentication domain), can be left empty if realm is not known.realm="example.net"###SIP favourite contacts (friends, buddies...)# Again it is represented as a family of [friend_X] sections.[friend_0]#SIP address of buddyurl=Alice #Policy for incoming SUBSCRIBEs for presence.

Linphone has inside a separation between the user interfaces and the core engine, allowing to create various kind of user interface on top of the same functionalities.

Its drawback is that it turns the call in a kind of automatic half-duplex mode, which makes impossible to interrupt the person who is talking.

[sound]#Enables echo limiter#this basically consists in lowering the mic input (in software)#when the speaker level is above a certain threshold#the attenuation is made proportionnal to the speaker detected levelecholimiter=0#el_speed parameter: gain changes are smoothed with a coefficent#el_speed is this coefficient.

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