siproxd_guide.sgml

这是一个C程序

</screen>
    </sect1>

    <!-- Chapter 6.3: GS BT-100 with Siproxd running "in front of" a NAT router -->
    <sect1 label="6.3">
      <?dbhtml filename="siproxd_guide_c6s3.html">
      <title>GS BT-100 with Siproxd running "in front of" a NAT router</title>
      <para>Scenario:</para>
<screen>
private IP address range             :          Internet
10.0.0.x                             :          (public IP address range)
                                     :
                                     :         foo.bar.org
+-------------+               +--------------+
!             !.10         .1 ! masquerading ! publicIP
! SIP UA      !---------------! NAT router   !------------>>  
! BT-100      !      !        !              !
+-------------+      !        +--------------+
                     !    eth0       :        ppp0
                     !               :       
                     !               :       
                eth0 !.2                     
              +-------------+                
              !   siproxd   !                
              !             !                
              +-------------+                
</screen>
      <para>Siproxd is running on 10.0.0.2. The masquerading NAT
        router (e.g. a ADSL NAT router that cannot run any user
        applications).</para>
      <para>siproxd.conf:</para>
<screen>
if_inbound  = eth0
if_outbound = eth0
host_outbound = foo.bar.org
hosts_allow_reg = 10.0.0.0/24
sip_listen_port = 5060
daemonize = 1
silence_log = 1
log_calls = 1
user = siproxd
registration_file = /var/lib/siproxd_registrations
pid_file = /var/run/siproxd/siproxd.pid
rtp_proxy_enable = 1
rtp_port_low  = 7070
rtp_port_high = 7089
rtp_timeout = 300
default_expires = 600
debug_level = 0
debug_port = 0
</screen>
      <para>NAT router configuration:</para>
<screen>
forward all incoming traffic on 5060/udp to 10.0.0.2
forward all incoming traffic from 7070/udp - 7089/udp to 10.0.0.2
</screen>
      <para>Phone configuration:</para>
<screen>
IP Address:             10.0.0.10
Subnet Mask:            255.255.255.0
Default Router: 10.0.0.1
DNS Server 1:           &lt;DNS Server of your Internet provider>
SIP Server:             proxy01.sipphone.com
Outbound Proxy: 10.0.0.2
SIP User ID:            1747669xxxx
Authenticate ID:        1747660xxxx
Authenticate Passwd:    *********
Name:                   Your Name Here
Use DNS SRV:            no
User ID is phone #:     no
Sip Registration:       yes
Unregister on reboot:no
Register expiration:    60
Early Dial:             no
local SIP port: 5060
local RTP port: 5004
Use random port:        yes
NAT traversal:  no
Use NAT IP:             &lt;empty>
Subscribe for MWI:      No
Send DTMF:              via RTP (RFC2833)
</screen>
    </sect1>


    <!-- Chapter 6.4: Transparent SIP Proxy -->
    <sect1 label="6.4">
      <?dbhtml filename="siproxd_guide_c6s4.html">
      <title>Transparent SIP Proxy</title>
      <para>Scenario:</para>
<screen>
private IP address range             :          Internet
10.0.0.x                             :          (public IP address range)
                                     :
                                     :         foo.bar.org
+-------------+               +--------------+
!             !.10         .1 ! masquerading ! publicIP
! SIP UA      !---------------! Firewall     !------------>>  
!             !               ! siproxd      !
+-------------+               +--------------+
                          eth0       :        ppp0
</screen>
      <para>You may have a SIP UA (Phone) that does not allow the
        specification of an outbound proxy. If siproxd is running
        on the masquerading router, the following configuration will
        do so called transparent proxying. The firewall will redirect
        outgoing SIP messages to siproxd, however the local Client
        is not aware of it.</para>
      <para>siproxd.conf:</para>
<screen>
if_inbound  = eth0
if_outbound = ppp0
hosts_allow_reg = 10.0.0.0/24
sip_listen_port = 5060
daemonize = 1
silence_log = 1
log_calls = 1
user = siproxd
registration_file = /var/lib/siproxd_registrations
pid_file = /var/run/siproxd/siproxd.pid
rtp_proxy_enable = 1
rtp_port_low  = 7010
rtp_port_high = 7019
rtp_timeout = 300
default_expires = 600
debug_level = 0
debug_port = 0
</screen>
      <para>Firewall configuration (iptables):</para>
<screen>
# redirect outgoing SIP traffic to siproxd (myself)
iptables -t nat -A PREROUTING -m udp -p udp -i eth0 \
                              --destination-port 5060     -j REDIRECT
# allow incoming SIP and RTP traffic
iptables -A INPUT -m udp -p udp -i ppp0 --dport 5060      -j ACCEPT
iptables -A INPUT -m udp -p udp -i ppp0 --dport 7070:7089 -j ACCEPT
</screen>
    </sect1>

    <!-- Chapter 6.5: Masquerading an Asterisk box -->
    <sect1 label="6.5">
      <?dbhtml filename="siproxd_guide_c6s5.html">
      <title>Masquerading an Asterisk box</title>
      <para>Scenario:</para>
<screen>
private IP address range             :          Internet
10.0.0.x                             :          (public IP address range)
                                     :
                                     :         foo.bar.org
+-------------+               +--------------+
!             !.10         .1 ! masquerading ! publicIP
! Asterisk    !---------------! Firewall     !------------>>  
!             !  SIP trunk    ! siproxd      !
+-------------+               +--------------+
  ! ! ! ! !               eth0       :        ppp0
..!.!.!.!.!.....
 extensions
 (local SIP clients)
</screen>
      <para>Siproxd can also be used to masquerade an Asterisk server.
        The Asterisk server will register itself as a SIP UA (Client)
        to an external SIP registrar. In this example this would be
        again sipphone.com. As Asterisk does not allow to specify an
        SIP outbound proxy we use the same setup for transparent proxying.
        The context values of the asterisk configuration probably must
        be adapted to fit your needs.</para>
      <para>siproxd.conf:</para>
<screen>
if_inbound  = eth0
if_outbound = ppp0
hosts_allow_reg = 10.0.0.0/24
sip_listen_port = 5060
daemonize = 1
silence_log = 1
log_calls = 1
user = siproxd
registration_file = /var/lib/siproxd_registrations
pid_file = /var/run/siproxd/siproxd.pid
rtp_proxy_enable = 1
rtp_port_low  = 7070
rtp_port_high = 7089
rtp_timeout = 300
default_expires = 600
debug_level = 0
debug_port = 0
</screen>
      <para>Firewall configuration (iptables):</para>
<screen>
# redirect outgoing SIP traffic to siproxd (myself)
iptables -t nat -A PREROUTING -m udp -p udp -i eth0 \
                     --source 10.0.0.11 --destination-port 5060 -j REDIRECT
# allow incoming SIP and RTP traffic
iptables -A INPUT -m udp -p udp -i ppp0 --dport 5060      -j ACCEPT
iptables -A INPUT -m udp -p udp -i ppp0 --dport 7070:7080 -j ACCEPT
</screen>
      <para>Asterisk configuration (SIP related part):</para>
      <para>Note: Very important are the fromuser and fromdomain
        keywords in the client section. They are required to have
        Asterisk send the correct From headers in SIP dialogs.
        The used Asterisk version is 'SVN-branch-1.4-r62331M'.</para>
      <para>With newer Asterisk versions, it is no longer required
        to have a separate REGISTER definition, this can be made
        implicit in the SIP trunk config.</para>
<screen>
; sip.conf:

[general]
port = 5060           ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0    ; Address to bind to (all addresses on machine)
context = from-sip-external ; Send unknown SIP callers to this context

useragent = PBX       ; NOTE: some providers (e.g sipcall.ch) do simply
                      ;       not work with the default "AsteriskPBX"
                      ;       UA String.

; Network Settings
nat=never
localnet = 10.0.0.0/24
domain = 10.0.0.10

; Codecs
disallow=all
allow=gsm       ; 13 Kbps
allow=ulaw      ; 64 Kbps
allow=alaw      ; 64 Kbps
autoframing = yes

; SIP Settings
canreinvite = no      ; important!

; the following are just my settings I use, however
; I dont' consider them critical
allowexternaldomains = yes
allowexternalinvites = yes
allowguest = yes
allowsubscribe = no
allowtransfer = yes
alwaysauthreject = no
autodomain = yes
callevents = no
compactheaders = no
dumphistory = no
g726nonstandard = no
ignoreregexpire = no
jbenable = no
jbforce = no
jblog = no
maxcallbitrate = 384
maxexpiry = 3600
minexpiry = 180
notifyringing = no
pedantic = no
promiscredir = no
recordhistory = no
relaxdtmf = no
rtcachefriends = no
rtsavesysname = no
rtupdate = no
sendrpid = yes
sipdebug = no
t1min = 100
progressinband = no
;register = 
t38pt_udptl = no
trustrpid = no
usereqphone = no
videosupport = no
</screen>
      <para>The Trunk definition looks like:</para>
<screen>
; users.conf:

[general]
;
; Full name of a user
;
fullname = New User
userbase = 200
;
; Create voicemail mailbox and use use macro-stdexten
;
hasvoicemail = yes
;
; Set voicemail mailbox 6000 password to 1234
;
vmsecret = 1234
;
; Create SIP Peer
;
hassip = yes
hasiax = no
;
; Create H.323 friend
;
;hash323 = yes
;
; Create manager entry
;
hasmanager = no
;
; Remaining options are not specific to users.conf entries but are general.
;
callwaiting = yes
threewaycalling = yes
callwaitingcallerid = yes
transfer = yes
canpark = yes
cancallforward = yes
callreturn = yes
callgroup = 1
pickupgroup = 1
host = dynamic
localextenlength = 3
allow_aliasextns = no
allow_an_extns = no
hasagent = no
hasdirectory = no


;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
; Local SIP UAs
; = locally connected phones. nothing special here.
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
[201]
callwaiting = yes
cid_number = 201
context = local_sip
email = e@mail
fullname = Full Name
group = 
hasagent = yes
hasdirectory = yes
hasiax = no
hasmanager = no
hassip = yes
hasvoicemail = yes
host = dynamic
mailbox = 201
secret = sip_password
threewaycalling = yes
zapchan = 
registeriax = no
registersip = yes
vmsecret = 1234


;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
; SIP Trunks
; these are masqueraded via siproxd
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
[trunk_1]
disallow = all
allow = gsm,ulaw,alaw,adpcm,speex,g729,g723
callerid = 
contact = 17476691234       ; IMPORTANT
context = DID_trunk_1
dialformat = ${EXTEN:1}
fromdomain = proxy01.sipphone.com
fromuser = 17476691234      ; IMPORTANT
group = 
hasexten = no
hasiax = no
hassip = yes
host = proxy01.sipphone.com
insecure = very
port = 5060
provider = 
registeriax = no
registersip = yes
secret = sip_password
trunkname = Custom - sipphone1234
trunkstyle = customvoip
username = 17476691234
</screen>
    </sect1>
  </chapter>
</book>