Dokumen - Tips - Sip Trunking Manual
Dokumen - Tips - Sip Trunking Manual
Dokumen - Tips - Sip Trunking Manual
05.15
Nothing contained in this manual shall be deemed to be, and this manual does not constitute, a warranty of, or
representation with respect to, any of the equipment covered. This manual is subject to change without notice and
NEC Unified Solutions, Inc. has no obligation to provide any updates or corrections to this manual. Further, NEC
Unified Solutions, Inc. also reserves the right, without prior notice, to make changes in equipment design or
components as it deems appropriate. No representation is made that this manual is complete or accurate in all
respects and NEC Unified Solutions, Inc. shall not be liable for any errors or omissions. In no event shall NEC Unified
Solutions, Inc. be liable for any incidental or consequential damages in connection with the use of this manual. This
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VoIP
VoIP (Voice over Internet Protocol or Voice over IP) allows the delivery of voice information using the Internet protocol (send-
ing data over the Internet using an IP address). This means that voice information, in a digital form, can be sent in packets over
the Internet rather than using the traditional public switch telephone network (CO lines). A major advantage of VoIP and Inter-
net telephony is that it avoids the tolls charged by ordinary telephone service.
Using VoIP equipment at a gateway (a network point that acts as an entrance to another network), the packetized voice trans-
missions from users within the company are received and routed to other parts of the company’s intranet (local area or wide
area network) or they can be sent over the Internet using CO lines to another gateway. 1
IP Networking
IP Networking uses VoIP technology to connect two or more telephone systems together. This allows calls to be made between
sites without using the public telephone network. This can save a considerable amount of money, and can make communica-
tion between sites much easier.
Aspire Networking can only be implemented when all nodes are Aspire systems - if non-Aspire systems are to be networked,
H.323 or SIP trunks must be used.
SIP Trunking
SIP (Session Initiation Protocol) is a protocol used for Voice over IP. It is deÞned by the IETF (Internet Engineering Task
Force) in RFC2543 and RFC3261 (RFC3261 requires system software 5.10 or higher). SIP trunking is the term used for link-
ing a PBX, like the Aspire, to the public telephone network by means of VoIP. This provides the possibility for users to place
and receive communications and services from any location and for networks to identify the users wherever they are located.
SIP analyzes requests from clients and retrieves responses from servers then sets call parameters at either end of the communi-
cation, handles call transfer and termination. The Aspire implementation and programming for SIP and H.323 are very similar.
The call routing, call features and speech handling (RTP) are the same - only the signalling protocol is different.
With the Aspire, SIP trunks can receive incoming calls with Caller ID, place outgoing calls, and transfer SIP trunks to IP, SIP,
analog and digital stations, and across a network.
1. The voice quality of VoIP is dependent on variables such as available bandwidth, network latency and Quality of
Service (QoS) initiatives, all of which are controlled by the network and internet service providers. Because these
variables are not in NEC’s control, it cannot guarantee the performance of the user’s IP-based remote voice solu-
tion. Therefore, NEC recommends connecting VoIP equipment through a local area network using a Private IP
address.
Conditions
● If entries are made in Program 10-29-xx for a SIP Server and the SIP Server is then removed or not used, the
entries in Program 10-29-xx must be set back to their default settings. Even if Program 10-29-01 : SIP
Proxy Setup - Outbound Proxy is set to “0” (off), the Aspire will still check the settings in the remaining
10-29 programs.
● The Aspire does not support the simultaneous use of a SIP trunk inter-connection and a SIP trunk carrier
connection.
● The Aspire restricts an outgoing call under the following conditions:
1. SIP conÞguration failed
2. SIP registration failed
3. NTCPU/VOIPU link down
4. Lack of VOIPU DSP resource
5. Lack of bandwidth
● Aspire SIP does not support T.38 FAX (H.323 trunk supports T.38 FAX).
SIP Setup
1. By default, the Aspire is assigned a static IP address and runs behind a NAT router.
When using an Aspire on a LAN behind a NAPT router, forward port 5060 to the IP address of the
Aspire NTCPU (since the signaling is handled by the NTCPU). Then, since the media stream (the
speech) uses a large range of ports for the RTP packages, forward the ports (10020 through 10083) to
the IP address of the VoIP PCB. Or, use the DMZ option for the VOIPU PCB. This means that the
VOIPU PCB is not actually behind the Þrewall. This is achieved by connecting the VOIPU PCB to a
physical or virtual DMZ port. You can also acheive the same result by port forwarding 10020 to 10083.
2. DeÞne the SIP Carrier account information (user name, password, domain name/IP address to the provider).
3. DeÞne the trunk ports as SIP (and not H.323).
4. Set the Expire Time.
Protocols
● SIP (RFC2543 bis04)
● SIP RFC3261 Supported [updated version of RFC2543] (requires software 5.10 or higher)
The SIP stack has been updated from RFC2543 Base to RFC3261 Base.
● SDP
● RTP/RTCP
● UDP
● IPv4
With system software 5.10 or higher, the following enhancements are available:
● Support the 401 response for the Initial Invite
If 401 message is sent for the Initial Invite, with previous software, the system can not respond to the mes-
sage correctly.
● Support the 401/407 response for the Invite of Session Timer
If 401/407 response is sent for the invite of Session Timer, the system can send the Invite message with
Authentication header.
● Support the 128 byte size of Nonce max value sent by the 401/407 message
The Nonce max size sent by the 401/407 message is expanded to 128 byte. With previous software, 64 byte
size can be received.
Authentication Process
● When using an external SIP Server provided by a carrier, usually an authentication process is needed.
● Aspire SIP trunks support HTTP digest authentication process (MD5).
● This process is done on a Register process and Initial INVITE process.
Caller ID
● Caller ID for SIP Trunks is set by Program 21-17 : IP (H.323/SIP) Trunk Calling Party Number Setup
for Trunks.
● Caller ID for SIP Extensions is Program 21-19 : IP (SIP) Trunk Calling Party Number Setup for
Extensions.
● Programs follow program priority as follows: 21-19 > 21-17 > 10-28-04
● With a trunk-to-trunk transfer and Trunk-to-Trunk Outgoing Caller ID Through Mode enabled (Program
14-01-24), the Caller ID/sub address received from the incoming trunk will be sent. Prior to 3.xx software,
the system would use the Caller ID set in Program 21-12 (ISDN) or Program 21-17 (IP). If a SIP trunk is
connected to a SIP carrier, then the sub-address is not transferred.
Carrier Support
● If a common carrier supports SIP, then the Aspire can connect the SIP Carrier, the outgoing call to the PSTN
network and the common IP network via an Aspire SIP trunk.
● A conformity test with a carrier’s SIP server is recommended.
Early Media
● When the Aspire receives the response 18x w/SDP and the codec negotiation is a success, the Aspire starts to
send/receive RTP packets.
Fault Tolerance
● When Aspire uses an external SIP Server, if the registration process fails, the Aspire blocks outgoing calls
with SIP trunks. All SIP trunk ports are placed into a busy status.
● If the Aspire has trunk groups which includes both SIP trunks and ISDN trunks, when all SIP trunks are
busy, a user can make an outgoing call using an ISDN trunk as a bypass.
QoS
● Aspire SIP trunks support TOS (Program 84-10 : ToS Setup, protocol type=6 (SIP Trunk).
Registration Process
● When Aspire registers its own IDs with a external SIP server, the following system data is sent:
Authentication Authentication
Register ID # User ID ID Password
Register ID 0 10-28-04 10-30-02 10-30-03
Register ID 1 10-36-02 10-36-03 10-36-04
: 10-36-02 10-36-03 10-36-04
:
Register ID 31 10-36-02 10-36-03 10-36-04
● The Aspire sends the REGISTER Message when the system starts up, register timer expires, NTCPU LAN
links and recover timer expires.
IP DSP Resource
➻ 10-19-01 : VOIPU DSP Resource Selection
Select the type of VOIPU DSP Resource. This program setting has no affect on the terminal/trunk port assign-
ments or usage.
Entries: 0=common (ICM/Trunks), 1=extension, 2=trunk, 3=networking (Default: 0=Common)
SIP Caller ID
➻ 14-01-24 : Basic Trunk Data Setup - Trunk-to-Trunk Outgoing Caller ID Through Mode
Enable or disable the Trunk-to-Trunk Outgoing Caller ID Through Mode. This option allows Caller ID from the
original outside caller to be displayed when a trunk is forwarded off premise. This option can only be used with
PRI and SIP trunks.
Entries: 0=Disabled, 1=Enabled (Default: 0)
➻ 21-17-01 : IP Trunk Calling Party Number Setup for Trunk
This program assigns the Caller Party Number for each IP trunk. The assigned number is sent to the central ofÞce
when the caller places an outgoing call. If the Calling Party Number is assigned by both 21-17 and 21-18/21-19,
then the system uses the entry in 21-18/21-19.
Entries: 1-0, *, # (Default: None)
➻ 21-19-01 : IP Trunk Calling Party Number Setup for Extension
This program is used to assign the Calling Party Number for each extension. The assigned number is sent to the
central ofÞce when the caller places an outgoing call. If the Calling Party Number is assigned by both Program
21-17 and 21-18/21-19, then the system uses the data in Program 21-18/21-19.
Entries: 1-0, *, # (Default: None)
SIP CODEC
➻ 84-13-01 : SIP Trunk CODEC Information Basic Setup - G.711 Audio Frame Number
Set the G.711 Audio Frame Number.
Entries: 1-3 (Default: 2)
➻ 84-13-02 : SIP Trunk CODEC Information Basic Setup - G.711 VAD Detection Mode
Enable or disable the G.711 VAD Detection Mode.
Entries: 0=Disable, 1=Enable (Default: 0)
➻ 84-13-03 : SIP Trunk CODEC Information Basic Setup - G.711 Type
DeÞne the G.711 type.
Entries: 0=A-law, 1=μ-law (Default: 1)
➻ 84-13-04 : SIP Trunk CODEC Information Basic Setup - G.711 Jitter Buffer (Min)
Set the minimum G.711 Jitter Buffer.
Entries: 0-145 ms (Default: 30 ms)
SIP Protocol
➻ 84-14-01 : SIP Trunk Basic Information Setup - INVITE ReTx Count
Set the INVITE Re TX Count.
Entries: 0-255 (Default: 7)
➻ 84-14-02 : SIP Trunk Basic Information Setup - Request ReTX Count
Set the Request Re TX Count.
Entries: 0-255 (Default: 11)
➻ 84-14-03 : SIP Trunk Basic Information Setup - Response ReTX Count
Set the Response Re TX Count.
Entries: 0-255 (Default: 7)
➻ 84-14-04 : SIP Trunk Basic Information Setup - Request ReTX Start Time
Set the Request Re TX Start Time.
Entries: 0-65535 (0ms-6553.5 seconds) (Default: 5 (500ms))
SIP UPnP
➻ 10-37-01 : UPnP Setup - UPnP Mode
Use this option to determine whether UPnP task starts. If UPnP task starts, it obtains a NAPT router WAN IP
Address by using NAT traversal and saves it in 10-12-07 automatically.
Entries: 0=Disable, 1=Enable (Default: 0)
➻ 10-37-02 : UPnP Setup - UPnP Interval
UPnP task will try to obtain the WAN IP Address of the NAPT router at the interval deÞned in this option.
Entries: 0-3600 (Default: 60)
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NEC UniÞed Solutions, Inc.
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(0893214) TEL: 800-365-1928 FAX: 203-926-5458 April 19, 2006
www.necuniÞedsolutions.com Printed in U.S.A.