Experimental VoIP service for research purposes

SIP Proxy and registrar: SER 2.0
Freeswitch from svn
Asterisk 1.2
SER status page

TODO

HA
Presence
IPv6
Install SEMS

DNS nameserver configuration (bind)

		; See http://www.ietf.org/rfc/rfc3263.txt		

		; NAPTR (Naming Authority Pointer)
		@               IN      NAPTR   0 0 "s" "SIPS+D2T" "" _sips._tcp
		@               IN      NAPTR   1 0 "s" "SIP+D2T"  "" _sip._tcp
		@               IN      NAPTR   2 0 "s" "SIP+D2U"  "" _sip._udp

		; SRV (Service)
		_sip._udp       IN      SRV 0 0 5060    sip
		_sip._tcp       IN      SRV 0 0 5060    sip
		_sips._tcp      IN      SRV 0 0 5061    sip

		_stun._udp      IN      SRV 0 0 3478    kursk
		_stun._tcp      IN      SRV 0 0 3478    kursk
		;_stun-pass._tls        IN      SRV 0 0 3478    kursk

Standard service ports

port 5050 - SIP Proxy and Registrar (Repro)
port 5060 - SIP Proxy and Registrar (SER)
port 5070 - Media server (Asterisk)
port 5080 - [reserved for outbound proxy]
port 5090 - Hi-Fi media server (Freeswitch)

Recommended SIP useragents

X-lite (Windows, Mac, Linux)
ekiga for linux. Supports video and messaging
Twinkle for linux. Supports Speex Ultra-Wideband (32 kHz)
GAIM. Messaging and presence
kphone for linux. Supports NAPTR and SRV
linphone for linux. Supports Speex 16 kHz
Nokia E61
Snom 300

Useragent configuration

X-lite:

Go to System Settings, SIP account, Default:

		Username:           your username
		Authorization user: your username
		Password:           your secret password
		Domain/Realm:       your domain e.g. symbianos.org

In Presence, change mode to "Presence Agent"

Nokia/Symbian-phones

If you are behind SIP ALGs, it is better to use TCP.

ProfileName=SymbianOS
ServiceProfile=IETF
DefaultAccessPoint=ContractInternet
PublicUserName=sip:username@symbianos.org
UseCompression=No
Registration=AlwaysOn
UseSecurity=No
ProxyServerAddr=sip:symbianos.org
Realm=symbianos.org
UserName=username
Password=password
AllowLooseRouting=Yes
TransportType=UDP
Port=5060
RegistrarServerAddr=sip:symbianos.org
Realm=symbianos.org
UserName=username
Password=password
TransportType=UDP
Port=5060

NAT traversal

STUN attributes

IPv6

IPv6 tunneling: TSPC and MIREDO

Streaming bridge requirements

Streaming source in any format (OGG, MP3, ...)
It must be possible to specify several sources per instance
It must be possible to specify several sources per useragent (for redundancy)
SIP-end should handle several active calls
SIP-end should handle multiple audio/video codecs

SIP software

- asterisk	GPL
- Ekiga 	GPL
- eXosip2	GPL+Proprietary
- FreeSwitch	MPL
- kphone	GPL
- linphone	GPL
- minisip 	LGPL+GPL
- opal		MPL
- openser	GPL
- opensipstack	MPL
- osip2		GPL+Commercial
- pjsip		GPL
- resiprocate	Vovida licence (BSD-like)
- ser		GPL+Proprietary
- Sofia		LGPL
- twinkle	GPL
- Yate 		GPL
- X-lite 	(Proprietary/Closed source)