Quickstart guide for RTClite SIP Caller. Setup, settings, how-to. Place VoIP/SIP calls from Python. Includes example template.

rtclite_sip_caller.py

# -*- coding: utf-8 -*-

from rtclite.app.sip.caller import Caller, default_options
from rtclite.std.ietf.rfc2396 import Address, URI
from rtclite.std.ietf import rfc4566
rfc4566.lineending = '\n'

import logging
logging.basicConfig(level=logging.INFO)


'''
QUICK SETUP GUIDE AND SIMPLIFIED USAGE
Installing and importing RTClite SIP Caller into Python as a module.
See end of this file for an easy template.
-------------------------------------------------------------------

Make sure you're using Python 2.7.
Verify by typing: python --version


Type:

git clone https://github.com/theintencity/rtclite.git
git clone https://github.com/theintencity/py-audio.git
sudo pip install gevent

IF you have problem installing gevent due to the greenlet dependency, you may want to install greenlet explicitly as follows, before installing gevent:
sudo easy_install greenlet==0.4.10


Now, we must add sip caller to PYTHONPATH.
On Ubuntu we will add the $PYTHONPATH to our .bashrc profile.

Type (replace the path with the path of your installation):
cp ~/.bashrc ~/.bashrc.bak && echo 'export PYTHONPATH="${PYTHONPATH}:/home/av1d/rtclite"' >> ~/.bashrc
(.bashrc.bak becomes your profile backup just in case).

Now reload .bashrc by typing:
source ~/.bashrc

The module will now be working from within Python.
'''



# All of these options are options of the command line software.
# Type: python -m rtclite.app.sip.caller -h
# in terminal to get all the options or see below for complete list.
# They're accessed by "options.whatever_option_you_want".
# Follow the same syntax as below (replace hyphen with underscore, etc.):


options = default_options()
options.to = Address('sip:18007741500@tollfree.alcazarnetworks.com')
options.uri = URI('sip:18007741500@tollfree.alcazarnetworks.com')
options.samplerate = 41000
options.domain = 'example.net'
options.use_lf = True

caller = Caller(options)
try: caller.wait()
except: pass
caller.close()


'''
Options:
  -h, --help            show this help message and exit
  -v, --verbose         enable verbose mode for this module
  -q, --quiet           enable quiet mode with only critical logging
  --test                run any tests and exit

  Network:
    Use these options for network configuration

    --int-ip=INT_IP     listening IP address for SIP and RTP. Use this option
                        only if you wish to select one out of multiple IP
                        interfaces. Default "0.0.0.0"
    --ext-ip=EXT_IP     IP address to advertise in SIP/SDP. Use this to
                        specify external IP if running on EC2. Default is to
                        use "--int-ip" if supplied or any local interface,
                        which is "127.0.1.1"
    --transport=TRANSPORTS
                        the transport type is one of "udp", "tcp" or "tls".
                        Default is "udp"
    --port=PORT         listening port number for SIP UDP/TCP. TLS is one more
                        than this. Default is 5092
    --listen-queue=LISTEN_QUEUE
                        listen queue for TCP socket. Default is 5
    --max-size=MAX_SIZE
                        size of received socket data. Default is 4096
    --fix-nat           enable fixing NAT IP address in Contact and SDP

  SIP:
    Use these options for SIP configuration

    --user-agent=USER_AGENT
                        set this as User-Agent header in outbound SIP request.
                        Default is empty "" to not set
    --subject=SUBJECT   set this as Subject header in outbound SIP request.
                        Default is empty "" to not set
    --user=USER         username to use in my SIP URI and contacts. Default is
                        "av1d"
    --domain=DOMAIN     domain portion of my SIP URI. Default is to use local
                        hostname, which is "superscape"
    --proxy=PROXY       IP address of the SIP proxy to use. Default is empty
                        "" to mean disable outbound proxy
    --authuser=AUTHUSER
                        username to use for authentication. Default is to not
                        use authentication
    --authpass=AUTHPASS
                        password to use for authentication. Only used together
                        with --authuser
    --strict-route      use strict routing instead of default loose routing
                        when proxy option is specified
    --to=TO             the target SIP address, e.g., '"Henry Sinnreich"
                        <sip:henry@iptel.org>'. This is mandatory
    --uri=URI           the target request-URI, e.g., "sip:henry@iptel.org".
                        Default is to derive from the --to option
    --listen            enable listen mode with or without REGISTER and wait
                        for incoming INVITE or MESSAGE
    --register          send REGISTER to SIP server. This is used with
                        --listen to receive incoming INVITE or MESSAGE and
                        with --to if the SIP server requires registration for
                        outbound calls
    --register-interval=REGISTER_INTERVAL
                        registration refresh interval in seconds. Default is
                        3600
    --retry-interval=RETRY_INTERVAL
                        retry interval in seconds to re-try if register or
                        subscribe fails. Default is 60
    --send=SEND         enable outbound instant message. The supplied text
                        with this option is sent in outbound MESSAGE request
    --auto-respond=AUTO_RESPOND
                        automatically respond to an incoming INVITE or MESSAGE
                        if we are not already in a call. Default is 200 to
                        auto accept. Use 0 to not respond
    --auto-respond-after=AUTO_RESPOND_AFTER
                        number of seconds after which to auto-respond an
                        incoming call if we are available. Default is 3
    --auto-terminate-after=AUTO_TERMINATE_AFTER
                        number of seconds after which to auto-terminate an
                        accepted incoming call. Default is 0 to not auto-
                        terminate
    --use-lf            use LF as line ending instead of default CRLF

  Media:
    Use these options for media configuration

    --no-sdp            disable sending SDP in outbound INVITE
    --no-audio          disable audio in a call
    --audio-loopback    enable audio loopback mode where this agent sends back
                        the received audio to the other end
    --samplerate=SAMPLERATE
                        audio samplerate for capture and playback. Default is
                        44100. Use 48000 on OS X
    --no-touchtone      disable sending touch tone (DTMF) digits when typed.
                        By default touch tone sending is enabled
    --recognize         enable speech recognition to show the received audio
                        as text
    --textspeech        enable text to speech to send typed text as audio. If
                        touchtone is also enabled, it distinguishes between
                        typed digits and non-digits to determine whether to
                        use touchtone or textspeech for the typed text.
'''



# QUICK SETUP TEMPLATE - PASTE INTO ITS OWN FILE (remove the ''' at beginning/end):


'''
# -*- coding: utf-8 -*-

from rtclite.app.sip.caller import Caller, default_options
from rtclite.std.ietf.rfc2396 import Address, URI
from rtclite.std.ietf import rfc4566
rfc4566.lineending = '\n'

import logging
logging.basicConfig(level=logging.INFO)

options = default_options()
options.to = Address('sip:18007741500@tollfree.alcazarnetworks.com')
options.uri = URI('sip:18007741500@tollfree.alcazarnetworks.com')
options.samplerate = 41000
options.domain = 'example.net'
options.use_lf = True
#options.int_ip =
#options.ext_ip =
#options.transport =
#options.port =
#options.listen_queue =
#options.max_size =
#options.fix_nat
#options.user_agent =
#options.subject =
#options.user =
#options.proxy =
#options.authuser =
#options.authpass =
#options.strict_route
#options.listen
#options.register
#options.register_interval =
#options.retry_interval =
#options.send =
#options.auto_respond =
#options.auto_respond_after =
#options.auto_terminate_after =
#options.no_sdp =
#options.no_audio =
#options.audio_loopback =
#options.no_touchtone
#options.recognize
#options.textspeech


caller = Caller(options)
try: caller.wait()
except: pass
caller.close()
'''