September 2011
05-2666-004
www.dialogic.com
Dialogic® 1000 and 2000 Media Gateway Series
SIP Compliance (Version 6.0 SU8 Software)
Copyright and Legal Notice
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Dialogic® 1000 and 2000 Media Gateway Series SIP Compliance
Table of Contents
1. Scope ............................................................................................................. 7
2. References ..................................................................................................... 8
3. Configuration Parameters .............................................................................. 9
Configuration INI Parameters ........................................................................... 9
Configuration XML parameters ........................................................................ 14
4. General ........................................................................................................ 19
VOIP to TDM Connect Determination ............................................................... 19
SDP Usage ................................................................................................... 19
5. Functions ..................................................................................................... 20
6. Supported RFCs and Drafts .......................................................................... 21
7. Methods ....................................................................................................... 23
8. Responses .................................................................................................... 24
1xx Response – Information Responses ........................................................... 24
2xx Response – Successful Responses ............................................................. 24
3xx Response – Redirection Responses ............................................................ 24
4xx Response – Request Failure Responses ...................................................... 25
5xx Response – Server Failure Responses ........................................................ 26
6xx Response – Global Responses ................................................................... 27
9. HeaderS ....................................................................................................... 28
10. SIP METHODS .............................................................................................. 31
INVITE Requests (Generated) ......................................................................... 31
From Header ............................................................................................. 31
Contact Header .......................................................................................... 31
To Header ................................................................................................. 32
Diversion Header........................................................................................ 32
Route Header ............................................................................................ 33
INVITE Requests (Received) ........................................................................... 34
Vendor-Specific Port Specifier ...................................................................... 34
Determination of Number to Dial .................................................................. 34
Early-Media ............................................................................................... 35
Positive Answering Machine Detection (PAMD) ............................................... 37
INVITE Responses ......................................................................................... 38
INFO ........................................................................................................... 40
BYE ............................................................................................................. 42
CANCEL ....................................................................................................... 43
OPTIONS ..................................................................................................... 43
Proxy-Server Monitoring ............................................................................. 43
VoIP Endpoint Monitoring‟ ........................................................................... 44
NOTIFY (“message-summary”) ....................................................................... 45
REFER ......................................................................................................... 46
REGISTER .................................................................................................... 47
11. Call Flows .................................................................................................... 49
VOIP to TDM Success .................................................................................... 49
VOIP to TDM Success .................................................................................... 50
VOIP to TDM Success (ISDN CONNECT) ........................................................... 51
VOIP to TDM Success (ISDN Progress Indication:8 In-Band)............................... 52
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Dialogic® 1000 and 2000 Media Gateway Series SIP Compliance
VOIP to TDM Success with Early Media ............................................................ 53
VOIP to TDM Failure – Call Un-Routable ........................................................... 54
VOIP to TDM Failure – Call Canceled ................................................................ 55
VOIP to TDM Failure – TDM Channel Unavailable ............................................... 56
VOIP to TDM Failure – Glare ........................................................................... 57
VOIP to TDM Failure – Busy Response ............................................................. 58
VOIP to TDM Failure – Error Response ............................................................. 59
TDM to VOIP Success (Post-INVITE CPID) ........................................................ 60
TDM to VOIP Success (Calling Number Updated) ............................................... 61
TDM to VOIP Success with Early Media (Gateway Model) .................................... 62
TDM to VOIP Success with Early Media (Application Model) ................................ 63
TDM to VOIP Failure - Rejected ....................................................................... 64
TDM to VOIP Failure - TDM Cancel................................................................... 65
VOIP to VOIP Success – Redirect Routing ......................................................... 66
VOIP to VOIP Success – Bridged Routing with Pass Through ............................... 67
VOIP to VOIP Success – Bridged Routing with Transcoding ................................. 69
VOIP to VOIP Failure – Bridged Routing with Pass Through ................................. 70
Hold, Unhold ................................................................................................ 71
VOIP Call Drop .............................................................................................. 72
TDM Call Drop .............................................................................................. 73
Unsupervised Transfer Success - VOIP Target - REFER....................................... 74
Unsupervised Transfer Failure - VOIP Target - REFER ........................................ 75
Unsupervised Transfer – TDM Target ............................................................... 76
Unsupervised Transfer Success - TDM Target - REFER .................................... 76
Unsupervised Transfer Success - TDM Target - REFER .................................... 77
Unsupervised Transfer Failure - TDM Target – REFER Error Response ............... 78
Unsupervised Transfer Failure - TDM Target – REFER Busy Response ................ 79
Supervised Transfer Success - VOIP Target, Gateway is Transferee ..................... 80
Supervised Transfer Success - VOIP Target, Gateway is Transferor ..................... 81
Supervised Transfer Success - TDM Target, Gateway is Transferee and Transferor 82
Supervised Transfer Success - TDM Target, Gateway is Target ........................... 83
Supervised Transfer – TDM Target, Gateway is Transferee and Target ................. 84
Supervised Transfer Success - TDM Target, Gateway is Transferee and Target ..... 85
Supervised Transfer Failure - TDM Target – Canceled ..................................... 86
Supervised Transfer Failure - TDM Target – Canceled ..................................... 87
Supervised Transfer Failure - TDM Target – Busy Response ............................. 88
Supervised Transfer Failure - TDM Target – Error Response ............................. 89
Supervised Transfer Failure - TDM Target Drops............................................. 90
Supervised Transfer Failure - TDM Transferee Drops ....................................... 91
MWI - VOIP to TDM Success ........................................................................... 92
MWI - VOIP to TDM Failure ............................................................................. 93
MWI - VOIP to TDM Failure - Glare .................................................................. 94
MWI – TDM to VOIP Success .......................................................................... 95
T.38 Fax - TDM to VOIP ................................................................................. 96
T.38 Fax – VOIP to TDM ................................................................................. 98
12. PROXY MONITORING ................................................................................. 101
13. SIP Authentication ..................................................................................... 103
Overview ................................................................................................... 103
Example WWW-Authenticate Header ............................................................. 104
Example Authorization Header ...................................................................... 105
INVITE Authentication Success – Gateway is UAC ........................................... 106
INVITE Authentication Failure – Gateway is UAC ............................................. 107
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Dialogic® 1000 and 2000 Media Gateway Series SIP Compliance
INVITE Authentication Success – Gateway is UAS ........................................... 108
INVITE Authentication Failure – Gateway is UAS ............................................. 109
Register Authentication Success – Gateway is UAC .......................................... 110
14. Calling ID Privacy ...................................................................................... 111
Gateway Configuration ................................................................................ 112
Translation ................................................................................................. 113
SIP INVITE to ISDN SETUP ........................................................................ 113
ISDN SETUP to SIP INVITE ........................................................................ 115
15. User To User Information .......................................................................... 117
SIP to SIP calls ........................................................................................... 117
SIP to TDM and TDM to SIP calls ................................................................... 117
Transfers using SIP REFER ........................................................................... 119
Supported Encoding .................................................................................... 120
String Encoding........................................................................................ 120
Hex Encoding........................................................................................... 120
Comparison of String verse Hex Encoding ................................................... 120
16. Multipart MIME to encapsulate PSTN signaling .......................................... 121
ISDN Information Elements .......................................................................... 121
Information Element 77 (hex) .................................................................... 121
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Dialogic® 1000 and 2000 Media Gateway Series SIP Compliance
Revision
Release date
Notes
05-2666-004
September 2011
Version 6.0 SU8 Software
Last modified: September 2011
Revision History
Refer to www.dialogic.com for product updates and for information about support policies,
warranty information, and service offerings.
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Dialogic® 1000 and 2000 Media Gateway Series SIP ComplianceDocSubtitle
1. Scope
This document describes the implementation and usage of the Session Initiated Protocol (SIP) by
the Dialogic® 1000 Media Gateway Series (DMG1000) and Dialogic® 2000 Media Gateway Series
(DMG2000) product lines, which also are referred to collectively herein as Dialogic® Media
Gateway or Media Gateway or gateway. A gateway can provide a connection between VOIP
telephony networks and proprietary digital telephony networks.
This document assumes that the reader understands the function and operation of the gateway
(see the Dialogic® 1000 and 2000 Media Gateway Series User‟s Guide) and is familiar with SIP.
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Dialogic® 1000 and 2000 Media Gateway Series SIP ComplianceDocSubtitle
2. References
Dialogic® 1000 and 2000 Media Gateway Series User‟s Guide
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Dialogic® 1000 and 2000 Media Gateway Series SIP ComplianceDocSubtitle
Parameter (INI)
Valid Settings
Default
Description
dspDigitRelay
None, RFC2833,
Inband-Tone
RFC2833
RTP Digit Relay Mode.
dspFaxModemToneRelay
RFC2833, Inband-Tone
RFC2833
RTP Fax/Modem Tone Relay Mode.
dspLBRCodec
G.723.1, G.729AB
G.723.1
Selects the Low Bit Rate Codec.
dspVAD
On, Off
On
Voice Activity Detection.
gwFaxTransportMode
T.38,
G.711 Passthrough,
None
T.38
Defines fax-transport method.
gwIsdnIe77Enable
Yes, No
No
When enabled (set to „Yes‟), the
Gateway will propagate a received ISDN
information element 77h to a SIP
INVITE message. The data is sent as a
section of a multipart MIME message.
The information element is propagated
only when it is received in an ISDN
SETUP message.
gwMonitorCallConns
Yes, No
No
When enabled (set to 'Yes'), the Media
Gateway will monitor the connection
3. Configuration Parameters
The following table lists the configuration parameters that govern the operation of the SIP mode of the gateway. This is not a
complete list of all gateway configurable parameters.
See the Dialogic® 1000 and 2000 Media Gateway Series User‟s Guide for a list that was complete as of its date of publication.
Configuration INI Parameters
The following SIP configurations are stored in text format in the gateway file named config.ini.
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Dialogic® 1000 and 2000 Media Gateway Series SIP ComplianceDocSubtitle
Parameter (INI)
Valid Settings
Default
Description
state of active IP calls. If the active IP
call has lost connection, the Media
Gateway will tear down the call.
gwMonitorCallIntSec
10-3600
30
Call Monitor Interval (secs).
gwMonitorVoipHostsIntSec
10-3600
30
Host Monitor Interval (secs).
gwProactiveMonitorDnsARecordEnable
Yes, No
No
When the Proactive DNS Monitoring is
enabled (set to 'Yes'), the Gateway will
monitor the IP addresses obtained from
DNS resolution. If an IP address does
not respond to SIP OPTIONS pings, the
gateway does not attempt to send it
SIP INVITES. When the IP address
resumes responding to SIP OPTIONS
pings, the gateway resumes sending it
SIP INVITES.
gwQosCallControl
0-255
0
Call Control QoS Byte.
gwQosRtp
0-255
0
RTP QoS Byte.
gwRTPEndPort
6000-65000
50000
RTP Start Port.
gwRTPStartPort
6000-65000
49000
RTP Start Port.
gwRTPValidateSrcIp
On, Off
Off
RTP Source IP Address Validation
gwRTPValidateSrcPort
On, Off
Off
RTP Source UDP Port Validation.
gwSigDigitRelay
On, Off
Off
Signaling Digit Relay Mode.
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Dialogic® 1000 and 2000 Media Gateway Series SIP ComplianceDocSubtitle
Parameter (INI)
Valid Settings
Default
Description
gwU2UEnable
On, Off
Off
Enables/disables gateway support for
the user-user header. See chapter 15.
gwU2UTranslateMethod
String, Hex
String
If the user-user header does not specify
an encoding, this parameter is used as
the implied encoding.
secSipTlsProtocol
SSLv3_TLSv1,
SSLv3_Only,
TLSv1_Only
SSLv3_TLSv1
SSL TLS Protocol.
secSipTlsUseSelfSignedCert
Self-Signed,
CA-Signed
Self-Signed
TLS Certificate Type.
sipAcceptableMedia
RTP_Only,
SRTP_Only,
RTP_SRTP
RTP_SRTP
Acceptable Media on INVITE reception.
sipCallAsDomainName
No, Yes
No
Defines the host name used in the From
header of generated INVITEs. If 'Yes',
the gateway's SIP Server Domain is
used as the host name. If 'No', the
gateway's VOIP address is used as the
host name.
sipDnsServerAddr
IPv4 dotted decimal
address.
Blank
IP Address in dotted decimal notation.
sipDnsServerAddr2
IPv4 dotted decimal
address.
Blank
IP Address in dotted decimal notation
(secondary).
sipEarlyMediaSupport
Always,
On-Demand,
None
OnDemand
RFC 3960 Early Media Support.
sipEnumDnsEnabled
Yes, No
No
DNS Translation of Phone Numbers.
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Dialogic® 1000 and 2000 Media Gateway Series SIP ComplianceDocSubtitle
Parameter (INI)
Valid Settings
Default
Description
sipExpInvSec
0-60000
120
Invite Expiration Time (secs).
sipPrivacyHdrEnabled
Yes, No
No
Enable SIP Privacy headers (RFC3325 &
RFC3323, and draft-ietf-sip-privacy-04
sipPrivacyHdrMethod
P-Asserted-Identity
Remote-Party-ID
Both
P-AssertedIdentity
Supported headers
sipRelProvRsp
None,
Required,
Supported
Supported
Reliable Provisional Responses.
sipRetryAfter
1–60000
60
SIP Retry After hint (sec).
sipSendSupportedSDPMedia
Yes, No
No
Send Supported SDP Media Types.
sipServerDomain
String with length
between
1-254 characters.
pbxgw.default.com
Host and domain name.
sipServerPort
1024-65000
5060
TCP/UDP Server Port.
sipSipsUriEnabled
Yes, No
No
SIPS URI Scheme.
sipT1Multiplier
1-255
64
T1 Multiplier.
sipT1TimeMs
100-60000
500
T1 timer (msecs) – Request Retransmit
Start Time.
sipT2TimeMs
200-60000
(must be greater than
T1)
4000
T2 timer (msecs) – Request Retransmit
Max Time.
sipT4TimeMs
1000-60000
5000
T4 timer. (msecs) – Specifies amount of
time the network will take to clear
messages between client and server
transactions.
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Dialogic® 1000 and 2000 Media Gateway Series SIP ComplianceDocSubtitle
Parameter (INI)
Valid Settings
Default
Description
sipTcpInactivitySec
10-60000
90
Number of seconds after which an idle
TCP connection will be closed.
sipTlsCertVerifyDate
Yes, No
Yes
Verify TLS Peer Certificate Date.
sipTlsCertVerifyPurpose
Yes, No
Yes
Verify TLS Peer Certificate Purpose.
sipTlsCertVerifyTrust
Yes, No
Yes
Verify TLS Peer Certificate Trust.
sipTlsCipherListType
RSA,
RSA_NULL_ENCRYPTION
RSA
Type of cipher list used by SIP TLS.
sipTlsEnabled
Yes, No
No
TLS Transport Enabled.
sipTlsInactivitySec
10-60000
30
TLS Inactivity Timer (sec).
sipTlsMutualAuthentication
Yes, No
Yes
Mutual TLS Authentication Required.
sipTlsServerPort
1024-65000
5061
TLS Server Port.
sipUdpTcpEnabled
Yes, No
Yes
UDP/TCP Transports Enabled.
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Dialogic® 1000 and 2000 Media Gateway Series SIP ComplianceDocSubtitle
Parameter (XML)
Valid Settings
Default
Description
RouteTable/
VoipHostGroups/
Group/Host
Valid endpoint
address (VOIP, URL,
alias, or blank)
Blank
Specifies a VOIP endpoint that is to receive calls from the either the
TDM or VOIP network.
RouteTable/
VoipHostGroups/
Group/FaultTolerant
Yes, No
No
Enables/Disables fault-tolerant handling of outbound VOIP calls.
If 'Yes', then the Media Gateway will failover to the next configured
VOIP endpoint if an outbound VOIP call attempt fails.
If 'No', then a failed outbound VOIP call attempt will not be retried.
RouteTable/
VoipHostGroups/
Group/LoadBalance
Yes, No
No
Enables/Disables load-balancing of outbound VOIP calls.
If 'Yes', then the Media Gateway will route outbound VOIP calls to
each configured VOIP endpoint in a round-robin fashion.
If 'No', then the Media Gateway will not load-balance across multiple
VOIP endpoints.
SIP/
NetworkGroups/
Group/
Signaling/
Transport
TCP, UDP, TLS
UDP
Preferred Call Signal transport to be used on all initial requests sent
by the gateway.
SIP/
Yes, No
No
Applicable only if 'Transport Type' is configured for TLS.
Configuration XML parameters
The following SIP configuration items are stored in XML format in the gateway file named dmg.xml.
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Dialogic® 1000 and 2000 Media Gateway Series SIP ComplianceDocSubtitle
Parameter (XML)
Valid Settings
Default
Description
NetworkGroups/
Group/
Signaling/
SIPS
Yes = All Request, To, From, and Contact URIs generated by the
gateway will use the SIPS URI scheme.
No = All Request, To, From, and Contact URIs generated by the
gateway will use the SIP URI scheme.
SIP/NetworkGroups/Grou
p/Signaling/UserPhone
Yes, No
Yes
Specifies if the user=phone parameter is specified in the SIP URI for
TDM calls.
SIP/NetworkGroups/Grou
p/Signaling/
PhoneContextLocal
String with length
between
1–128 characters.
Blank
Specifies the phone-context of the SIP URI on the From: line. If
blank, then no phone-context is specified.
SIP/NetworkGroups/Grou
p/Signaling/
PhoneContextRemote
String with length
between
1–128 characters.
Blank
Specifies the phone-context of the SIP URI on the To: line. If blank,
then no phone-context is specified.
SIP/NetworkGroups/Grou
p/Proxy/Server/Addr
String with length
between
1–128 characters.
Blank
The SIP URI of the Primary Proxy Server through which the Gateway
may send/receive requests. If blank, the Gateway
will not use the Primary Proxy Server.
SIP/NetworkGroups/Grou
p/Proxy/Server/Port
1024-65000
5060
The IP Port of the SIP Proxy Server.
SIP/NetworkGroups/Grou
p/Proxy/Query
10-36000
30
Interval at which the Proxy Server(s) are queried.
The Proxy Server must respond to a SIP OPTIONS request in order
for the proxy query to succeed.
If the Primary Proxy Server does not respond to the query, the
gateway switches to the Backup Proxy Server.
Once the Primary Proxy Server responds to the query, the gateway
switches back to the Primary Proxy Server.
SIP/NetworkGroups/Grou
String with length
Blank
IP Address of the SIP Registration Server that the Gateway should
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Dialogic® 1000 and 2000 Media Gateway Series SIP ComplianceDocSubtitle
Parameter (XML)
Valid Settings
Default
Description
p/Register/Addr
between
1–128 characters.
register with. If blank, the Gateway will not
register with a Registration Server.
SIP/NetworkGroups/Grou
p/Register/Port
1024-65000
5060
IP Port of the SIP Registration Server.
SIP/NetworkGroups/Grou
p/Register/User
String with length
between
1-64 characters.
Blank
Specifies the User-Field of the address-of-record to be registered. If
blank, then no User-Field is specified.
SIP/NetworkGroups/Grou
p/Register/GwName
String with length
between
1-64 characters.
Blank
Specifies the Gateway Name in the SIP Register message. If blank
the gateway‟s IP address is used.
SIP/NetworkGroups/Grou
p/Register/Expire
10-60000
120
Specifies the value to be placed in the Expires header of transmitted
INVITE requests. If the value is zero, an Expires header is not added
to the INVITE request.
SIP/NetworkGroups/Grou
p/Audio/Codec
G.711u,
G.711a,
G.723.1,
G.729AB
G.711u,
G.711a
Audio codec preference selection.
SIP/NetworkGroups/Grou
p/Audio/G711PacketSiz
e
10,20, 30
30
Size of a G.711 codec audio packet. Packet size = (Frame Size *
Frames Per Packet).
SIP/NetworkGroups/Grou
p/Audio/
G723PacketSize
30, 60
30
Size of a G.723 codec audio packet. Packet size = (Frame Size *
Frames Per Packet)
SIP/NetworkGroups/Grou
p/Audio/
G729PacketSize
10,20,30,40,50,60
30
Size of a G.729 codec audio packet. Packet size = (Frame Size *
Frames Per Packet).
SIP/NetworkGroups/Grou
p/SRTP/Preference
None, RTP_Only,
SRTP_Only
RTP_Only
Determines how SRTP is negotiated for a media session.
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Dialogic® 1000 and 2000 Media Gateway Series SIP ComplianceDocSubtitle
Parameter (XML)
Valid Settings
Default
Description
SIP/NetworkGroups/Grou
p/SRTP/AuthTag
32, 80
80
Length of the SHA1 authentication tag in bits.
SIP/NetworkGroups/Grou
p/SRTP/TxMKI
Yes, No
Yes
If enabled then a one-byte MKI (Master Key Index) is used when
transmitting secure-RTP and secure-RTPC packets.
If disabled, MKI is not used.
SIP/NetworkGroups/Grou
p/SRTP/KDREnable
Yes, No
Yes
If enabled then a new encryption key is derived after 2^KDR
packets. If disabled, a key is derived only at the start of the
transmission.
SIP/NetworkGroups/Grou
p/SRTP/KDR
16-24
16
Key derivation rate for outbound INVITE‟S. This value is the
exponent of a power of 2.
SIP/NetworkGroups/Grou
p/SRTP/WSH
64-99
64
Anti-replay window size hint.
SIP/NetworkGroups/Grou
p/SRTP/UnEncryptSRTP
Yes, No
No
If enabled, then transmitted RTP VOICE packets will NOT be
encrypted despite the negotiation of cipher keys.
SIP/NetworkGroups/Grou
p/SRTP/UnEncryptSRTC
P
Yes, No
No
If enabled, then transmitted RTCP CONTROL packets will NOT be
encrypted despite the negotiation of cipher keys.
SIP/NetworkGroups/Grou
p/SRTP/UnAuthSRTP
Yes, No
No
If enabled, then RTP VOICE packets will NOT be authenticated.
SIPAuth/Authentication
/Enabled
Yes, No
No
Enables Authentication for Inbound SIP Calls.
SIPAuth/Authentication
/GwRealm
String with length
between
1-64 characters.
default.gw.
com
Sets the Gateway Realm for Inbound SIP calls.
SIPAuth/Authentication/
Users/User/UserName
String with length
between
1-64 characters.
Blank
Specifies an acceptable User Name associated with GwRealm for
inbound SIP methods.
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Dialogic® 1000 and 2000 Media Gateway Series SIP ComplianceDocSubtitle
Parameter (XML)
Valid Settings
Default
Description
SIPAuth/Authentication/
Users/User/Password
String with length
between
1-64 characters.
Blank
Specifies the Password associated with UserName for inbound SIP
methods.
SIPAuth/Authentication
/Algorithm
MD5,
MD5-sess
MD5
Sets the Authentication algorithm.
SIPAuth/Authentication/
Methods/Method/Name
Invite, Register,
Notify, Info, Bye,
Refer, Options
N/A
Identifies the SIP method which may be Authenticated. There is one
entry for each SIP method (valid setting).
SIPAuth/Authentication/
Methods/Method/
Challenge
Yes, No
No
Determines if the specified SIP method (Name) will be
Authenticated on receive. There is one entry for each SIP method
(valid setting).
SIPAuth/Authorization
/Enabled
Yes, No
No
Enables Authorization for outbound SIP methods.
SIPAuth/Authorization
/Realms/RealmName
String with length
between
1-64 characters.
Blank
Defines the Realm to be associated with outbound SIP methods.
SIPAuth/Authorization
/Realms/UserName
String with length
between
1-64 characters.
Blank
Defines the User Name to be associated with RealmName for
outbound SIP methods.
SIPAuth/Authorization
/Realms/Password
String with length
between
1-64 characters.
Blank
Defines the Password to be associated with UserName for outbound
SIP methods.
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4. General
VOIP to TDM Connect Determination
The gateway does not send the 200 OK response to the INVITE until a positive detect event is
detected on the TDM network call. The positive detect event could be voice activity, ring-back
tone termination, etc. In this case, the gateway waits for the TDM network call to be answered
before the 200 OK response is sent and the bi-directional media channel is established. If the
gateway detects error or busy tone on the TDM network call during call origination, then the
gateway terminates the TDM network call and responds with a failure to the INVITE request. This
is generally preferred if the gateway is to be used by a voice-mail system or other 'automated‟
system that requires positive answer verification on the TDM network.
SDP Usage
The gateway supports G.711, G.723.1 and G.729AB codecs. The codec preference is set in the
gateway‟s configuration.
The SDP information placed in INVITE requests reflects the configured codec preference. The RTP
channels used by the gateway are also configurable. When SRTP is supported, SDP INVITE also
contains the media type, either RTP/SAVP for secured or RTP/AVP for non-secured. For RTP/SAVP
media type, crypto information is provided including encryption algorithm, authentication type and
tag, master Key Index (MKI), key derivation rate (KDR), and window size hint (WSH).
On INVITE requests generated by the gateway, the gateway sends the desired RTP transmit port
as well as the codec preferences. The gateway expects the 200 OK response from the far party to
contain the far party‟s transmit RTP port as well as its codec preference. The gateway will use the
first codec specified in the far party‟s preference list that matches a codec in the gateway‟s codec
preference list. If no match is found, then the gateway disconnects the call. This should not
happen since the far party should not send a 200 OK if there are no matching codec preferences.
On INVITE requests received by the gateway, the gateway expects the SDP information to include
the requesting party‟s codec preferences. If none of the preferences match the gateway‟s
capabilities, or the INVITE request does not specify an RTP port, then the gateway responds with
a 488 Not Acceptable error. The gateway will use the first codec specified in the far party‟s
preference list that matches a codec in the gateway‟s codec preference list. If no SDP is included
in the INVITE, the gateway may respond with a 200 OK which contains the gateway‟s SDP offer.
A VOIP hold request is performed by the gateway by using the a=sendonly attribute. The gateway
will accept a hold request from a peer through either the a=sendonly attribute or via setting
in the SDP of a re-INVITE request (to which the gateway will respond with an
a=sendonly attribute). The gateway stops all RTP transmit for the call. The un-hold request is
handled by setting in the SDP of a re-INVITE request or by removing the
a=sendonly attribute. The gateway continues RTP transmit for the call.
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Dialogic® 1000 and 2000 Media Gateway Series SIP ComplianceDocSubtitle
Function
Supported?
User Agent Client (UAC)
Yes
User Agent Server (UAS)
Yes
Proxy Server
3rd Party
Redirect Server
Yes
5. Functions
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Specification
Notes
RFC 3261 SIP
None
RFC 3263 DNS Resolution
None
RFC 2976 SIP INFO Method
None
RFC 3824 Using ENUM for SIP Applications
None
RFC 4028: The SIP Session Timer
None
RFC 3265: SIP – Specific Event Notification
None
RFC 3515: The REFER Method
Used only for call transfer.
Draft-ietf-sipping-cc-transfer-07: Call Transfer
None
RFC 3892: The Referred-By Mechanism
None
RFC 3891: The SIP Replaces Header
None
RFC 3264: An Offer Answer Model with Session
Description
None
ITU-T Recommendation T.38: Procedures for real-time
Group 3 facsimile communication over IP networks
Only UDPTL transport is
supported.
RFC 3842: A Message Summary and Message Waiting
Indication Event Package for the Session Initiation
Protocol
None
RFC 3326: The Reason Header for the Session Initiation
Protocol
None
RFC 3959: The Early Session Disposition Type for the
Session Initiation Protocol
Application model
support. No support for
Gateway model.
RFC 3960: Early Media and Ringing Tone Generation in
the Session Initiation Protocol (SIP)
Application model
support. No support for
Gateway model.
RFC 3262: Reliability of Provisional Responses in
Session Initiation Protocol (SIP)
Supports reliable
provisional responses and
reception of PRACK, but
does require them.
RFC 4568: Security Descriptions for Media Streams
None
RFC 3398: ISDN ISUP to SIP Mapping
None
RFC 4566: Session Description Protocol
None
RFC 2617: HTTP Authentication
Used for SIP
Authentication. Basic
Authentication is not
supported for SIP. Digest
6. Supported RFCs and Drafts
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Dialogic® 1000 and 2000 Media Gateway Series SIP ComplianceDocSubtitle
Authentication is
supported
Draft-johnston-cuss-sip-uui-01.pdf
Using the transport
mechanism described in
section 3.6
22
7. Methods
Method
Supported?
Comments
INVITE
Yes
Generated and received for call initiation and
hold/unhold.
ACK
Yes
Generated and received.
OPTIONS
Yes
Generated to monitor status of proxy server and VOIP
endpoints (when
RouteTable/VoipHostGroups/Group/FaultTolerant
is enabled).
BYE
Yes
Generated and received.
INFO
Yes
Generated and received. Generated on TDM to VOIP call
party update.
CANCEL
Yes
Generated and received.
REGISTER
Yes
Generated for gateway registration and ignored if
received.
REFER
Yes
Generated and received for call transfer.
NOTIFY
Yes
Generated and received for transfer and MWI.
SUBSCRIBE
Yes
Received, but ignored.
MESSAGE
Yes
Received, but ignored.
PRACK
Yes
Generated if a 183 response is received with 100rel.
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Dialogic® 1000 and 2000 Media Gateway Series SIP ComplianceDocSubtitle
Response
Supported?
Comments
100 Trying
Yes
Generated for an incoming INVITE.
180 Ringing
Yes
Generated when ring-back tone is detected
on outbound TDM call.
181 Call is being forwarded
Yes
Not generated. Ignored on receive.
182 Queued
Yes
Not generated. Ignored on receive.
183 Session Progress
Yes
Generated and received. Generated in
response to early media request or call
progress from TDM side.
Response
Supported?
Comments
200 OK
Yes
Generated and received
202 Accepted
Yes
Generated and received (REFER, NOTIFY).
Response
Supported?
Comments
300 Multiple Choices
Yes
Not generated. If received, Contact list is
traversed.
301 Moved Permanently
Yes
Not generated. If received, Contact list is
traversed.
302 Moved Temporarily
Yes
Generated if inbound SIP call is to be
redirected to another IP address due to
dial plan rules. If received, Contact list is
traversed.
305 Use Proxy
Yes
Not generated. If received, Contact list is
traversed.
380 Alternate Service
Yes
Not generated. If received, call gracefully
fails/disconnects.
8. Responses
1xx Response – Information Responses
2xx Response – Successful Responses
3xx Response – Redirection Responses
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Dialogic® 1000 and 2000 Media Gateway Series SIP ComplianceDocSubtitle
Response
Supported?
Comments
400 Bad Request
Yes
Generated on receive of invalid requests.
When received, call gracefully
fails/disconnects.
401 Unauthorized
Yes
Generated when UAS Authentication is
enabled. When received, re-sends the
method with the Authorization header if
UAC Authentication is enabled.
402 Payment Required
Yes
Not generated. When received, call
gracefully fails/disconnects.
403 Forbidden
Yes
Generated on receive of INVITE that
cannot be routed to a telephony port.
When received, call gracefully
fails/disconnects.
404 Not Found
Yes
Generated when Special Information Tone
(SIT) is detected for reasons of "operator
intercept" or "vacant circuit" TDM.
Also generated if an MWI request to the
TDM network fails because of an unknown
extension number. When received, call
gracefully fails/disconnects.
405 Method Not Allowed
Yes
Generated on receive of a non-supported
method. When received, call gracefully
fails/disconnects.
406 Not Acceptable
Yes
Generated on receive of any method which
fails SIP authentication. When received,
call gracefully fails/disconnects.
407 Proxy Authentication
Required
Yes
Not generated. When received, re-sends
the method with the Authorization header
if UAC Authentication is enabled.
408 Request Timeout
Yes
Not generated. When received, call
gracefully fails/disconnects.
409 Conflict
Yes
Not generated. When received, call
gracefully fails/disconnects.
410 Gone
Yes
Not generated. When received, call
gracefully fails/disconnects.
411 Length Required
Yes
Not generated. When received, call
gracefully fails/disconnects.
413 Request Entity Too Large
Yes
Not generated. When received, call
gracefully fails/disconnects.
414 Request URL Too Long
Yes
Generated. When received, call gracefully
4xx Response – Request Failure Responses
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Dialogic® 1000 and 2000 Media Gateway Series SIP ComplianceDocSubtitle
Response
Supported?
Comments
fails/disconnects.
415 Unsupported Media
Yes
Not generated. When received, call
gracefully fails/disconnects.
420 Bad Extension
Yes
Not generated. When received, call
gracefully fails/disconnects.
480 Temporarily Unavailable
Yes
Generated when call can be routed to TDM
network, but the telephony port is already
active. Generated when outbound PSTN
call or PSTN MWI set/clear fails because of
glare condition. When received, call
gracefully fails/disconnects.
481 Call
Leg/Transaction/Subscription
Does Not Exist
Yes
Generated and received. When received,
call gracefully fails/disconnects.
482 Loop Detected
Yes
Not generated. When received, call
gracefully fails/disconnects.
483 Too Many Hops
Yes
Not generated. When received, call
gracefully fails/disconnects.
484 Address Incomplete
Yes
Not generated. When received, call
gracefully fails/disconnects.
485 Ambiguous
Yes
Not generated. When received, call
gracefully fails/disconnects.
486 Busy Here
Yes
Generated when busy tone detected on
TDM. When received, call gracefully
fails/disconnects.
487 Request Canceled
Yes
Generated upon receiving a CANCEL for a
pending INVITE. When received, call
gracefully disconnects.
488 Not Acceptable
Yes
Generated if a received INVITE contains
no supported media types. When received,
call gracefully fails/disconnects.
489 Bad Event
Yes
Generated if a NOTIFY is received for an
event that the gateway does not support.
When received, the corresponding NOTIFY
request fails.
Response
Supported?
Comments
500 Internal Server Error
Yes
Generated if internal Server error occurs.
When received, call gracefully
fails/disconnects.
5xx Response – Server Failure Responses
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Dialogic® 1000 and 2000 Media Gateway Series SIP ComplianceDocSubtitle
501 Not Implemented
Yes
Not generated. When received, call
gracefully fails/disconnects.
502 Bad Gateway
Yes
Not generated. When received, call
gracefully fails/disconnects.
503 Service Unavailable
Yes
Generated on detection of Special
Information Tone (SIT) for reasons of
"reorder" or "no circuit found" on TDM.
When received, call gracefully
fails/disconnects.
504 Gateway Timeout
Yes
Not generated. When received, call
gracefully fails/disconnects.
505 Version Not Supported
Yes
Not generated. When received, call
gracefully fails/disconnects.
513 Message Too Large
Yes
Not generated. When received, call
gracefully fails/disconnects.
Response
Supported?
Comments
600 Busy Everywhere
Yes
Not generated. When received, call
gracefully fails/disconnects.
603 Decline
Yes
Not generated. When received, call
gracefully fails/disconnects.
604 Does Not Exist Anywhere
Yes
Not generated. When received, call
gracefully fails/disconnects.
606 Not Acceptable
Yes
Not generated. When received, call
gracefully fails/disconnects.
6xx Response – Global Responses
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Dialogic® 1000 and 2000 Media Gateway Series SIP ComplianceDocSubtitle
Header
Supported?
Comments
Accept
Yes
Accept-Encoding
Yes
Accept-Language
Yes
Allow
Yes
Allow-Events
Yes
Authentication-Info
No
Also
No
Authorization
Yes
Generated
Call-ID
Yes
Generated
Contact
Yes
Generated
Content-Disposition
No
Content-Encoding
Yes
Generated
Content-Language
No
Content-Length
Yes
Generated
Content-Type
Yes
Generated
Cseq
Yes
Generated
Date
Yes
Diversion
Yes
Generated
Encryption
No
Error-Info
No
Event
Yes
Expires
Yes
Generated
From
Yes
Generated
In-Reply-To
No
Join
Yes
Generated
Max-Forwards
Yes
MIME-Version
No
Generated
Min-Expires
Yes
Organization
Yes
P-Asserted-Identity
Yes
Generated
9. Headers
All supported headers in the table are supported in the receive direction. Those generated as well
are noted.
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Dialogic® 1000 and 2000 Media Gateway Series SIP ComplianceDocSubtitle
Priority
Yes
Privacy
Yes
Generated
Proxy-Authenticate
Yes
Proxy-Authorization
Yes
Proxy-Require
Yes
Reason
Yes
Generated
Record-Route
Yes
Remote-Party-ID
Yes
Generated
Reply-To
No
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Dialogic® 1000 and 2000 Media Gateway Series SIP ComplianceDocSubtitle
Referred-By
Yes
Generated
Referred-To
Yes
Generated
Replaces
Yes
Generated
Requested-By
Yes
Generated
Require
Yes
Response-Key
No
Retry-After
Yes
Route
Yes
Generated
Server
Yes
Generated
Subject
Yes
Subscription-State
Yes
Supported
Yes
Timestamp
Yes
To
Yes
Generated
Unsupported
Yes
User-Agent
Yes
Generated
User-To-User
Yes
Generated
Via
Yes
Generated
Warning
Yes
WWW-Authenticate
Yes
Generated
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