I have been looking for a way to set up the Android SIP stack to be able to establish a SIP call between two devices on the same network, in an ad-hoc manner. i.e without REGIST
You can do this with CSipSimple, which is open source: http://code.google.com/p/csipsimple/
You set up local accounts, register to yourself instead of a server, then make a phone call using TXT mode and dial remote_account_name@remote_ip_address.
I have been stuck on the same problematic.
If you can make it without the android sip api, you can look at the rtp api which gives you a bit lower-level tools to make a P2P VOIP application without the need of a server.
To support audio conferencing and similar usages, you need to instantiate two classes as endpoints for the stream:
AudioStream specifies a remote endpoint and consists of network mapping and a configured AudioCodec. AudioGroup represents the local endpoint for one or more AudioStreams. The AudioGroup mixes all the AudioStreams and optionally interacts with the device speaker and the microphone at the same time.
The counterpart is that you have to write your own device discovery protocol in order to know the port used by the audiostream peer as explained in this answer
The problem is not so hard if you only intend to make one-to-one conversation but is a little bit trickier if you want to make one-to-n conversation.
For a one-to-n conversation, the conference host has to instanciate n audiostream for each remote device he wants to call. Each remote peer has only one audiostream linked to one of the host audiostream.
Sip peer is like an extension number used to configure in sip phone . Please find details for creating sip peer . I am using centos 6.9 64 bit and having installed asterisk 11 You can create sip peer using asterisk server . Goto vi /etc/asterisk/sip.conf
[1001]
username=1001
secret=123
qualify=yes
type=friend
disallow=all
allow=ulaw,alaw,gsm
host=dynamic
For more detail and easy understanding. Please refer given below link
https://youtu.be/27wm-fu25SM
or
http://rulariteducation.blogspot.in/2017/07/how-to-add-sip-peer-in-asterisk.html