ETSI TR 183 072 V3.1.1 (2010-09)

Technical Report
Telecommunications and Internet converged Services and
Protocols for Advanced Networking (TISPAN);
Emulation Services for PSTN Modem Calls

2 ETSI TR 183 072 V3.1.1 (2010-09)

Reference
DTR/TISPAN-03215-NGN-R3
Keywords
PSTN, SIP, H.248, interworking
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ETSI
3 ETSI TR 183 072 V3.1.1 (2010-09)
Contents
Intellectual Property Rights . 5
Foreword . 5
1 Scope . 6
1.1 Conventions . 6
1.1.1 SDP Offer/Answer protocol variants . 6
1.1.2 Configuration (Codec) list . 6
2 References . 6
2.1 Normative references . 6
2.2 Informative references . 7
3 Definitions and abbreviations . 9
3.1 Definitions . 9
3.2 Abbreviations . 9
3.2.1 Reference Points . 10
4 Problem Statement . 10
5 Example scenario as introduction . 11
6 XoIP bearer services for emulating PSTN modem calls - Existing technologies and
recommendations . . 13
6.1 Indication and Negotiation of Media Configuration(s) . 14
6.2 User plane - Handling of signal processing functions . 15
6.2.1 GW behaviour (by SIP VGW, H.248 MG) . 15
6.2.2 GW control of EC and JB (in SIP VGW, H.248 MG) . 15
6.2.2.1 H.248 MG . 15
6.2.2.1.1 Echo control . 15
6.2.2.1.2 Jitter buffer control . 15
6.2.2.2 SIP VGW . 15
6.2.2.2.1 Echo control . 15
6.2.2.2.2 Jitter buffer control . 15
6.2.3 Call control level: indication of EC towards "remote side" . 16
7 Media-type Configurations for PSTN modem calls . 16
7.1 Overview . 16
7.1.1 VBDoIP-type specific Configurations . 17
7.1.2 FoIP-type specific Configurations . 17
7.1.2.1 Major Configurations . 17
7.1.2.2 Signalled versus provisioned T.38 Configurations . 17
7.2 Preferred Configuration/Codec Lists (PCL) . 17
7.2.1 Basic principles . 17
7.2.2 Signalling aspects . 18
7.3 Interoperability check . 19
7.4 Grouping of multiple media configurations . 19
7.4.1 Introduction. 19
7.4.2 Signalling capabilities for grouped media configurations. 20
7.4.2.1 Call control: SIP/SDP . 20
7.4.2.2 Gateway control: H.248 . 20
7.4.3 Interworking aspects . 20
7.5 Declaration of dedicated media configurations . 20
7.5.1 Introduction. 20
7.5.2 Signalling capabilities for declaration of media configurations . 21
7.5.2.1 Call control: SIP/SDP . 21
7.5.2.2 Gateway control: H.248 . 21
7.6 Mode transitioning . 21
7.6.1 Introduction. 21
7.6.2 Mode transitioning behaviour . 21
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4 ETSI TR 183 072 V3.1.1 (2010-09)
7.6.3 Mode transitioning support for V.152 and T.38 emulation services . 22
8 Requirements analysis & recommendations . 23
9 Use cases . 24
9.1 Scenario #1 - IMS-based PES scenario, intra-IMS call between two SIP gateways . 24
9.1.1 Scenario #1.1 - Two SIP VGWs . 24
9.1.2 Scenario #1.2 - Two SIP AGCFs . 25
9.2 IMS & IMS-based PES scenarios, general intra-IMS call. 26
9.2.1 Scenario #2.1 - IMS & IMS-based PES scenario, general intra-IMS call . 26
9.2.2 Scenario #2.2 - IMS & IMS-based PES scenario, intra-IMS call, unsuccessful negotiation . 26
9.3 Scenario #3 - Inter-IMS call between two provider domains (IMS peering) . 27
9.4 Scenario #4 - IMS-PSTN UNI call . 28
9.4.1 Scenario #4bis - IMS-based PES to PSTN UNI (single IMS provider) . 28
9.5 Scenario #5 - IMS-PSTN NNI call . 28
9.6 Scenario #6 - IMS-PSTN (general) . 29
9.7 Scenario #7 - PES-IMS call between two provider domains . 29
9.8 Scenario #8 - PES only . 29
9.9 Scenario #9 - Others . 30
9.10 Summary - Use Cases vs Signalling Capabilities . 30
Annex A: Support of XoIP bearer services for PSTN modem calls: Inventory of
correspondent TISPAN specifications . 31
A.1 T.38 and V.152 call support in IMS releases? . 34
A.1.1 "Open" IMS position . 34
A.1.2 "Strict service control" IMS position . 34
A.1.3 Discussion . 34
Annex B: Support of Revised SDP Offer/Answer Syntax by 3GPP SIP Profile specifications . 35
Annex C: Discussion of SIP/SDP interworking by example scenarios . 37
C.1 Media grouping . 37
C.1.1 Example 1 - Two SIP VGWs with different SDP Capability Sets, but successful usage of RFC 3388
media grouping . 37
C.1.2 Example 2 - RFC 3388 deficiencies . 40
C.2 Example 3 - More SIP VGWs with different SDP Capability Sets . 42
Annex D: "SDP Offer/Answer protocol variants" - Negotiation Phases . 44
Annex E: Bibliography . 47
History . 48

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5 ETSI TR 183 072 V3.1.1 (2010-09)
Intellectual Property Rights
IPRs essential or potentially essential to the present document may have been declared to ETSI. The information
pertaining to these essential IPRs, if any, is publicly available for ETSI members and non-members, and can be found
in ETSI SR 000 314: "Intellectual Property Rights (IPRs); Essential, or potentially Essential, IPRs notified to ETSI in
respect of ETSI standards", which is available from the ETSI Secretariat. Latest updates are available on the ETSI Web
server (http://webapp.etsi.org/IPR/home.asp).
Pursuant to the ETSI IPR Policy, no investigation, including IPR searches, has been carried out by ETSI. No guarantee
can be given as to the existence of other IPRs not referenced in ETSI SR 000 314 (or the updates on the ETSI Web
server) which are, or may be, or may become, essential to the present document.
Foreword
This Technical Report (TR) has been produced by ETSI Technical Committee Telecommunications and Internet
converged Services and Protocols for Advanced Networking (TISPAN).
ETSI
6 ETSI TR 183 072 V3.1.1 (2010-09)
1 Scope
The present document specifically describes the framework of IP emulation services for PSTN modem call types. The
present document provides, inter alia, a description of:
• the problem statement;
• summary of IP bearer services;
• overview of media-type configurations; and
• a collection of example network use cases;
for PSTN modem calls.
1.1 Conventions
1.1.1 SDP Offer/Answer protocol variants
The present document provides example signalling syntax. There are two models for the Session Description
Protocol (SDP) concerning the indication and negotiation of media and transport capabilities:
• the name "legacy SDP Offer/Answer" indicates SDP Offer/Answer according RFC 3264 [i.1];
• the name "revised SDP Offer/Answer" indicates SDP Offer/Answer according RFC 5939 [i.4] and
RFC MediaCapNeg [i.5].
The two SDP Offer/Answer protocol variants differ in terms of supported SDP syntax, but also in terms of negotiation
logic on semantical level (see also Annex D).
1.1.2 Configuration (Codec) list
The term media configuration (briefly configuration) is more comprehensive than the term codec. A media
configuration covers typically media type, media format, all media format attributes, media transport stack, media
transport capacity and all associated parameter values.
The term configuration list represents consequently a list of configurations within the SDP media description block.
2 References
References are either specific (identified by date of publication and/or edition number or version number) or
non-specific. For specific references, only the cited version applies. For non-specific references, the latest version of the
reference document (including any amendments) applies.
Referenced documents which are not found to be publicly available in the expected location might be found at
http://docbox.etsi.org/Reference.
NOTE: While any hyperlinks included in this clause were valid at the time of publication ETSI cannot guarantee
their long term validity.
2.1 Normative references
The following referenced documents are necessary for the application of the present document.
Not applicable.
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7 ETSI TR 183 072 V3.1.1 (2010-09)
2.2 Informative references
The following referenced documents are not necessary for the application of the present document but they assist the
user with regard to a particular subject area.
[i.1] IETF RFC 3264 (2002): "An Offer/Answer Model with the Session Description Protocol (SDP)".
[i.2] IETF RFC 4040 (2005): "RTP Payload Format for a 64 kbit/s Transparent Call".
[i.3] IETF RFC 4733 (2006): "RTP Payload for DTMF Digits, Telephony Tones, and Telephony
Signals".
[i.4] IETF RFC 5939 (2010): "SDP Capabiliy Negotiation".
[i.5] Draft IETF RFC MediaCapNeg (201X): "SDP media capabilities Negotiation", {draft-ietf-
mmusic-sdp-media-capabilities}.
NOTE: MediaCapNeg is a draft version of IETF RFC, the reference will be updated when it is published
formally.
[i.6] Draft IETF RFC MiscCapNeg (201X): "Miscellaneous Capabilities Negotiation in the Session
Description Protocol (SDP)", {draft-ietf-mmusic-sdp-misc-cap}.
NOTE: MiscCapNeg is a draft version of IETF RFC, the reference will be updated when it is published formally.
[i.7] Draft IETF RFC ConnCapNeg (201X): "Session Description Protocol (SDP) Extension For
Setting Up Audio Media Streams Over Circuit-Switched Bearers In The Public Switched
Telephone Network (PSTN)", {draft-ietf-mmusic-sdp-cs}.
NOTE: ConnCapNeg is a draft version of IETF RFC, the reference will be updated when it is published formally.
[i.8] ITU-T Recommendation G.168: "Digital network echo cancellers".
[i.9] ITU-T Recommendation G.711: "Pulse code modulation (PCM) of voice frequencies".
[i.10] ITU-T Recommendation V.150.1: "Modem-over-IP networks: Procedures for the end-to-end
connection of V-series DCEs".
[i.11] ITU-T Recommendation V.151: "Procedures for end-to-end connection of analogue PSTN text
telephones over an IP network utilizing text relay".
[i.12] ITU-T Recommendation V.152: "Procedures for supporting voice-band data over IP networks".
[i.13] ITU-T Recommendation V.153: "Interworking between T.38 and V.152 using IP peering for
realtime facsimile services".
[i.14] ITU-T Recommendation T.38: "Procedures for real-time Group 3 facsimile communication over
IP networks".
[i.15] ETSI TS 123 228: "Digital cellular telecommunications system (Phase 2+); Universal Mobile
Telecommunications System (UMTS); LTE; IP Multimedia Subsystem (IMS); Stage 2 (3GPP
TS 23.228)".
[i.16] ETSI TS 124 229: "Digital cellular telecommunications system (Phase 2+); Universal Mobile
Telecommunications System (UMTS); LTE; IP multimedia call control protocol based on Session
Initiation Protocol (SIP) and Session Description Protocol (SDP) (3GPP TS 24.229)".
[i.17] ETSI TS 126 114: "Universal Mobile Telecommunications System (UMTS); LTE; IP Multimedia
Subsystem (IMS); Multimedia telephony; Media handling and interaction (3GPP TS 26.114)".
[i.18] ETSI TS 129 238: "Universal Mobile Telecommunications System (UMTS); LTE; Interconnection
Border Control Functions (IBCF) - Transition Gateway (TrGW) interface; Ix interface; Stage 3
(3GPP TS 29.238)".
[i.19] ETSI TS 129 332: "Universal Mobile Telecommunications System (UMTS); Media Gateway
Control Function (MGCF) - IM Media Gateway (IM-MGW); Mn interface (3GPP TS 29.332)".
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8 ETSI TR 183 072 V3.1.1 (2010-09)
[i.20] ETSI TS 182 012: "Telecommunications and Internet converged Services and Protocols for
Advanced Networking (TISPAN); IMS-based PSTN/ISDN Emulation Sub-system (PES);
Functional architecture".
[i.21] ETSI TS 183 002: "Telecommunications and Internet converged Services and Protocols for
Advanced Networking (TISPAN); H.248 Profile Version 3 for controlling Access and Residential
Gateways".
[i.22] ETSI TS 183 018: "Telecommunications and Internet converged Services and Protocols for
Advanced Networking (TISPAN); Resource and Admission Control: H.248 Profile Version 3 for
controlling Border Gateway Functions (BGF) in the Resource and Admission Control Subsystem
(RACS); Protocol specification".
[i.23] ETSI TS 183 036: "Telecommunications and Internet converged Services and Protocols for
Advanced Networking (TISPAN); ISDN/SIP interworking; Protocol specification".
[i.24] ETSI TS 183 043: "Telecommunications and Internet converged Services and Protocols for
Advanced Networking (TISPAN); IMS-based PSTN/ISDN Emulation; Stage 3 specification".
[i.25] ETSI ES 283 003: "Telecommunications and Internet converged Services and Protocols for
Advanced Networking (TISPAN); IP Multimedia Call Control Protocol based on Session
Initiation Protocol (SIP) and Session Description Protocol (SDP) Stage 3 [3GPP TS 24.229
[Release X], modified]".
[i.26] ETSI ES 283 049: "Telecommunications and Internet converged Services and Protocols for
Advanced Networking (TISPAN); H.248 Profile for controlling Trunking Media Gateways (TMG)
[Endorsement of 3GPP TS 29.332 (V7), modified]".
[i.27] ETSI TS 129 231: "Digital cellular telecommunications system (Phase 2+); Universal Mobile
Telecommunications System (UMTS); Application of SIP-I Protocols to Circuit Switched (CS)
core network architecture; Stage 3 (3GPP TS 29.231 version 9.0.0 Release 9)".
[i.28] IETF RFC 3388: "Grouping of Media Lines in the Session Description Protocol (SDP)".
[i.29] IETF RFC 3407: "Session Description Protocol (SDP) Simple Capability Declaration".
[i.30] Draft ITU-T Recommendation H.248.80: "Gateway control protocol: Usage of the revised SDP
offer/answer model with H.248".
NOTE: This ITU-T Recommendation is still in work, the reference will be updated when it is published formally.
[i.31] ITU-T Recommendation Q.115.0: "Protocols for the control of signal processing network elements
and functions".
[i.32] ETSI ES 283 012 (V2.1.1): "Telecommunications and Internet converged Services and Protocols
for Advanced Networking (TISPAN); Interworking; Trunking Gateway Control Procedures for
interworking between NGN and external CS networks [Endorsement of 3GPP TS 29.412 (R8)]".
[i.33] ITU-T Recommendation H.248.31: "Gateway control protocol: Adaptive jitter buffer package".
[i.34] IETF RFC 3362 (2002): "Real-time Facsimile (T.38) - image/t38 MIME Sub-type Registration".
[i.35] TISPAN Temporary Document 08TD241 (2005-09): "H.248 Trunking GW Profile - Voiceband
Data (VBD) Service".
[i.36] TISPAN Temporary Document ACWG3-H248 TD08 (2005-11): "H.248 Trunking GW Profile -
Progress on clause 5.17.2.3 VBD Procedures".
[i.37] ETSI TS 124 292 (V9.3.0): "Universal Mobile Telecommunications System (UMTS); LTE; IP
Multimedia Core Network subsystem Centralized Services; Stage 3 (3GPP TS 24.292 Release 9)".
[i.38] ETSI TS 129 163: "Digital cellular telecommunications system (Phase 2+); Universal Mobile
Telecommunications System (UMTS); LTE; Interworking between the IP Multimedia (IM) Core
Network (CN) subsystem and Circuit Switched (CS) networks (3GPP TS 29.163)".
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9 ETSI TR 183 072 V3.1.1 (2010-09)
3 Definitions and abbreviations
3.1 Definitions
For the purposes of the present document, the following terms and definitions apply:
PSTN modem call: voiceband data call originating/terminating in a PSTN domain
NOTE: The term voiceband data (VBD) is an umbrella term for all kind of teleservices which using a "data-
oriented transport" in the frequency band of the narrowband voice spectrum (which is a 3,1-kHz-band).
The data-oriented transport is realized by modem protocols (definition as in clause 3.13 of ITU-T
Recommendation V.152 [i.12]), as defined e.g. within the ITU-T V.x-series of Recommendations.
Teleservices may be categorized into three major applications areas: facsimile, text-based communication
and general data services.
XoIP emulation service (for PSTN modem calls): emulation service in IP networks, based on appropriated gateway
technologies for interworking voiceband data information between the PSTN and IP networks
NOTE: Example emulation services for the three main VBD application areas, which may be summarized as (by
using notation "application/transport"):
� Facsimile/modem: Gateway technologies for PSTN-to-IP interworking see e.g. ITU-T
Recommendation V.152 [i.12] for pass-through mode and ITU-T Recommendation T.38 [i.14] as
packet-relay mode;
� Text/modem: Gateway technologies for PSTN-to-IP interworking see e.g. ITU-T
Recommendation V.152 [i.12] for pass-through mode and ITU-T Recommendation V.151 [i.11] as
packet-relay mode; and
� Data/modem: Gateway technologies for PSTN-to-IP interworking see e.g. ITU-T
Recommendation V.152 [i.12] for pass-through mode and ITU-T Recommendation V.150.1 [i.10]
as packet-relay mode.
3.2 Abbreviations
For the purposes of the present document, the following abbreviations apply:
ACL Answered Configuration (Codec) List
AGCF Access Gateway Control Function
AGW Access GateWay
ARGW Access Residential media GateWay
AS Application Server
B2BUA Back-to-back User Agent (SIP)
BICC Bearer-Independent Call Control
CMD Clearmode
CMDoIP Clearmode over IP (RFC 4040 [i.2])
CSCS Call/Session Control Server
DSP Digital Signal Processor
EC Echo Canceller
FoIP Facsimile over IP (T.38)
G3FE Group 3 Facsimile Equipment
GW Gateway
IAF Internet-aware Fax device (T.38)
IBCF Interconnect Border Control Function
IMS IP Multimedia Subsystem
IP Internet Protocol
ISDN Integrated Services Digital Network
JBIG Joint Bi-level Image Experts Group (T.38 coding scheme)
MG Media Gateway (H.248)
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10 ETSI TR 183 072 V3.1.1 (2010-09)
MGC Media Gateway Controller (H.248)
MMR Modified Modified Read (T.38 coding scheme)
MoIP (Data) Modem over IP (V.150.1)
NCL Negotiated Configuration (Codec) List
NNI Network Network Interface
NTE Network Telephony Event
NTEoIP Network Telephony Event over IP (RFC 4733 [i.3])
O/A (SDP) Offer/Answer
OCL Offered Configuration (Codec) List
PBX Private Branch Exchange
PCL Preferred Configuration (Codec) List
PES PSTN Emulation Subsystem
PSTN Public Switched Telephone Network
QoS Quality of Service
RGW Residential GateWay
RTP Real Time Protocol
SCL Supported Configuration (Codec) List
SDP O/A SDP Offer/Answer
SDP Session Description Protocol
SIP Session Initiation Protocol
TGW Trunking GateWay
ToIP Text over IP (V.151)
TrGW Transition Gateway (3GPP)
UA User Agent (SIP)
UNI User Network Interface
VBD Voiceband Data
VBDoIP Voiceband Data over IP (V.152)
VGW Voice Gateway (SIP)
VoIP Voice over IP
XoIP X over IP (with 'X' as placeholder for "IP application protocol 'X'")
3.2.1 Reference Points
For the purposes of the present document, the following network reference points are referred:
• TISPAN specific reference points: Ia, P1.
• Common reference points of TISPAN and 3GPP: Gm, Ici, Nb, Nc, Mn, Mp, Mw.
See also clause 3.2 in TS 123 228 [i.15] with regards to the 3GPP defined reference points.
4 Problem Statement
There is a pending issue with voiceband data (VBD) calls in IMS, PES and IMS-based PES with regards to the
unambiguous indication and negotiation of possible bearer emulation services in IP networks. The term voiceband
data (VBD) originates from data traffic using a modem-based transport via the voice frequency spectrum of a PSTN
voice line. VBD calls represent all PSTN modem call types, i.e. the three main categories of fax/modem, text/modem
and data/modem calls, which are originating or terminating in a PSTN/ISDN domain.
The problem of the emulation of PSTN modem calls via IP network (thus, an IP bearer emulation service) is
well-known, because there are in the meanwhile experiences of approx. 10 years of network operation -, but this
difficulty is not yet solved for all cases. There are still ongoing issues notified in the daily process of VoIP network
operation. This challenge is controllable in specific scenarios like private VoIP networks (enterprise solutions), single-
vendor public VoIP networks or/and markets with sufficient information about all legacy modem techniques which are
still in service.
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11 ETSI TR 183 072 V3.1.1 (2010-09)
That specific rationale was/is the root cause why the problem was so far disregarded in existing TISPAN releases. Other
reasons coming from the different handling of modem calls in mobile and fixed networks, which preventing
straightforward solutions for converged NGN/IMS infrastructures (due to the different handling of a terminating
modem call in a mobile terminal). It was in the meanwhile also observed, that not every implementation of VBD IP
emulation services is fully compliant to the underlying standardized technologies, or/and sufficiently tested against the
universe of legacy, often market-specific, often still proprietary modem types. And there is also insufficient
commitment to feed back experiences in the standardization/maintenance process for enhancing the aimed technologies
(like e.g. V.15X-series, T.38, etc.), knowing that already a lot of PSTN experts and modem expertise is just gone ("and
that loss of knowhow implies a difficult and expensive reverse engineering process when trying to fix PSTN / modem
problems").
However, the technologies for VBD IP bearer emulation services may be not blamed for that situation, the problem of
PSTN modem calls in IMS & VoIP NGNs is primarily still an implementation and interoperability issue (see e.g. also
SIP Forum activities to fix fax/modem calls: FoIP task group http://www.sipforum.org/content/view/310/252/) between
PSTN-to-IP gateways (or IP endpoints). Any interoperability problem between two peering gateways should be firstly
addressed by:
1) sufficient and unambiguous specification of the aimed media configuration for a VBD IP emulation service;
and
2) the explicit declaration and negotiation between the involved parties.
Item #1 may be addressed by explicit usage of V.152 [i.12] or a packet relay technique like T.38 [i.14] for fax/modem
calls ("but then all T.38 parameters are provided as defined in T.38 [i.14]").
Item #2 relates to the negotiation (and re-negotiation) process on call/session control level, thus subject of SIP and the
embedded SDP Offer/Answer protocol in IMS and IMS-based PES networks. A sufficient SDP O/A negotiation is often
lacking today (due to the implicit assumption of partially/entirely expected media configurations (defaults) of the
remote side; or/and rudimentary support of SDP O/A).
The present document is not providing any explicit change indications for TISPAN specifications. The purpose of the
present document is rather a summary of the problem statement and the indication of possible changes in order to relax
this dilemma.
The problem may be introduced and studied at the simplest network scenario of a pure IMS, single domain
configuration, see next clause.
5 Example scenario as introduction
Figure 1 illustrates the example of a single network domain for IMS and IMS-based PES. The outlined problem should
be firstly solved for the simple case of such intra-IMS calls before looking at interconnection scenarios of multiple
domains or private-to-public configurations.
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12 ETSI TR 183 072 V3.1.1 (2010-09)
s1 SIP s2 SIP
(Application) (Application)
Server Server
(e.g.) SIP Mw
Voiceband data calls may
(e.g. 3GPP / ETSI (e.g. 3GPP / ETSI
TISPAN IMS) TISPAN IMS)
originate/terminate in the PSTN
SIP Gm
domain:
• Fax/modem (like G3FE call);
• Data/modem (like V.90 data call);
x1
• Text/modem (like V.18 text call)
SIP Voice
Gateway
PSTN
domain
(e.g. ETSI TISPAN
SIP VGW)
x2
SIP Voice
Gateway
PSTN
domain
(e.g. ETSI TISPAN
SIP VGW)
SIP terminals may originate/terminate:
y1
• T.38 FoIP (as T.38 endpoint) SIP
Terminal
But not:
• V.152 VBDoIP (“that’s a pure PSTN-to-
(e.g. IMS UE) Single, public IMS domain
IP GW standard”)
y2
SIP
Terminal
(e.g. IMS UE)
z1
SIP-PBX
Private IP
(e.g. SIP Forum)
domain
Figure 1: Mix of SIP VGWs (IMS-based PES) & SIP UEs (IMS)
A PSTN (access) domain is interconnected via SIP VGWs to the IMS domain in IMS-based PES solutions. PSTN
modem calls originate/terminate in that PSTN domains (e.g. a SIP signalling session from x1 to x2 via s1 and s2).
However, such a call may also terminate in an IMS UE (or even originate) if that IP terminal provides correspondent
capabilities for handling modem-based VBD services (e.g. a SIP session from x1 to y2 via s1 and s2).
The focus is rather on PSTN traffic and not on ISDN, - due to the typical ratio of PSTN:ISDN traffic (in favour of
PSTN) and the lacking capability of PSTN signalling with regards to the indication of a "PSTN bearer service", (see
note).
NOTE: ISDN is different, e.g. a call originating from a G3FE looks like a speech telephony service to the
network, but an ISDN fax terminal (G4FE) could provide an explicit indication for a fax service.
Further: there is typically following ratio between speech telephony and modem calls: "PSTN speech" >> "PSTN
modem". The PSTN modem calls may be furthermore classified in three types (note: such a categorization is also
justified by the availability of correspondent IP packet relay techniques (like T.38 FoIP, V.151 ToIP, V.150.1 MoIP)):
Figure 2 provides an example distribution for PSTN modem calls. Such a distribution is primarily driven by the
installed base of terminal equipment for the particular modem calls.
The (estimated) traffic profile of PSTN modem calls is significant for any IMS deployment. For instance, the SDP
Offerer is in charge of providing:
a) appropriate media configurations for all modem types; and
b) fall-back media configurations in case that specific IP bearer emulation services may not be supported by the
peering side.
ETSI
13 ETSI TR 183 072 V3.1.1 (2010-09)
ALL PSTN calls = “Voice calls”.
ANY PSTN “voice call” may be a “X/modem
call”.
ISDN may provide additionally circuit-mode
data, ISDN (G4FE) fax, …
90%
Fax
T
e
x
t
5%
D
a
Note to T.38 advocates:
t
a
⇒ 1-out-of-10 X/modem calls will fail if just
5%
T.38 is offered (and not any other option
for text & data modem)
⇒ If such a unsuccess rate is inaceptable,
then the SDP Offer must provide a VBD
alternative in addition to T.38
Example traffic profile for the mean distribution of
voiceband data (modem) calls:
„Fax/modem : Text/modem : Data/modem = 90 : 5 : 5"

Figure 2: Traffic Profile for PSTN Modem Calls (example)
6 XoIP bearer services for emulating PSTN modem
calls - Existing technologies and recommendations
There is a single "pass-through" technology defined, VBDoIP according ITU-T Recommendation V.152 [i.12], for all
three PSTN modem call categories. V.152 may be thus used in general, and should be also the fall-back option if a
packet-relay technique is not supported / not wished / etc.
ITU-T Recommendations T.38 [i.15], V.151 [i.12] and V.150.1 [i.11] are the correspondent packet relay methods for
FoIP, ToIP and MoIP. Media configurations for packet relay operations are much more expensive than VBDoIP V.152
due to the significant difference of required resources in terms of DSP processing time and memory for the
interworking function.
The focus may be therefore just on V.152 and T.38 for above outlined example traffic profile of PSTN modem calls
(e.g. such a distribution would not justify any implicit support of V.150.1 per se).
Any network solution needs also to consider pre-V.152 VBDoIP support because V.152 was just published during
TISPAN R1 timeframe, thus is not yet integral part of all R1 specifications. The mode may be labelled as "pseudo-
VBD" (pVBDoIP) because:
The SDP Offerer is unable to provide explicit signalling elements (like V.152) for the VBDoIP service, and
thus merging the "audio mode" and "VBD mode" in a single mode, called pseudo-VBD mode. Such a single
mode requires a configuration which allows to transport audio and VBD signals with the same media (codec)
configuration. pVBDoIP is therefore typically using G.711 [i.9] without silence suppression, without adaptive
Jitter Buffer control, without gain control, without noise reduction, … overlaid by a G.168 [i.8] compliant echo
canceller (EC).
Pseudo-VBD is further even proprietary in the context of SIP VGWs (due to a lacking specification of this gateway
mode in earlier TISPAN releases (R1, R2) like in contrary to some H.248 profiles; note: the situation improved with
TISPAN R3 by an explicit definition of a "non-V.152" mode in TS 183 043 R3 [i.24]).
ETSI
14 ETSI TR 183 072 V3.1.1 (2010-09)
6.1 Indication and Negotiation of Media Configuration(s)
The PSTN-IP gateways (here SIP VGW), (see note 1), of the example IMS domain support V.152 and T.38, and the
IMS UE could provide support for T.38 (as Internet-aware fax endpoint).
NOTE 1: Out of scope here: other SIP-controlled PSTN-IP gateways like SIP analog terminal adaptors (ATA) or
SIP PBXs.
All SIP UA entities involved in the SDP O/A process consequently support the SDP information elements for V.152
and T.38 for the explicit indication and negotiation of media configurations. (Again the note that a complete set of SDP
elements (all attributes) is needed as specified in T.38 [i.14], and not just a sub-set.)
Further, SDP O/A provides a legacy negotiation protocol according to RFC 3264 [i.1], but also enhancements by latest
IETF updates (sometimes called "revised SDP O/A"). The negotiation process could benefit from that extended
negotiation protocol, particularly if a decision between a packet-relay and pass-through mode is envisioned, or/and
V.152 would be considered in sub-modes (e.g. V.152 non-assured vs V.152 assured transport modes), or/and backward
compatibility to non-V.152 implementations needs to be solved, or/and support of multiple T.38 transport modes,
or/and support of multiple T.38 assurance levels (FEC, redundancy), or/and etc.
Multiple media configuration options and a negotiation via revised SDP O/A may be a significant step forward in
relaxing the interoperability problem in heterogeneous, multi-vendor deployment scenarios.
The IMS SIP profile is already providing support of revised SDP O/A protocol elements, see ES 283 003 [i.25] and the
underlying TS 124 229 [i.16] ("SDPCapNeg support since 3GPP R7 and MediaCapNeg since 3GPP R9").
NOTE 2: The full negotiation of media configurations for V.152, T.38, etc. via revised SDP O/A requires both:
RFC 5939 [i.4] and MediaCapNeg [i.5].
Table 1 on SDP support is derived from what precedes.
Table 1: SDP Negotiation Capability support versus SDP V.152/T.38 support -
Matrix of SIP device specific options
NNeeggootiatiattioionn CCaappaabbililititieiess
SSDDPP c coommppatibatibililitityy ooff
SuSupppporortt of of LegLegaaccyy SDSDPP O O//AA SuSuppppoorrtt o off RevReviisseedd SDSDPP O O//AA
SISIPP p prroodduucctt iimpmplelemmeennttaattiioonn
((RRFFCC 32 326464)) („(„SSDDPPCCaappNNeegg““,, „ „MMedediiaaCCaappNNeegg““,, . .).)
SDSDPP
NN
NN foforr
SDSDPP
YY
T.T.3838
forfor
SDSDPP
NN
VV.152152
YY foforr
YY
T.T.3838
The SDP capabilities of a particular
implementation needs to be considered by a
service provider concerning his codec
preferrences.
The SDP capabilities affecting the
construction of an OCL and negotiation.

The outlined status of supported SDP capabilities by a SIP UA and B2BUA instance in an IMS domain is considered in
the SDP O/A negotiation process.
Note to packet relay methods (like T.38 FoIP):
The VBD packet relay traffic flow may either:
a) re-use the IP transport connection of the voice flow ("replaced voice"); or
b) use a separate, dedicated IP transport connection ("muted voice").
ETSI
15 ETSI TR 183 072 V3.1.1 (2010-09)
Both approaches are justified from application perspective (i.e. there might be different PSTN teleservices like a
"fax-only call" versus an "alternate speech-fax call").
However, independent of the selected transport method, an explicit indication and negotiation between both parties is
necessary.
6.2 User plane - Handling of signal processing functions
There are a number of signal processing functions with respect to the "voiceband" traffic: echo control (echo
cancellation, echo suppression), noise reduction, automatic level (gain) control, etc., which may be located at the border
of PSTN-IP networks, embedded in VoIP gateways (see clause 6.2 in ITU-T Recommendation Q.115.0 [i.31]).
Such kind of voiceband related signal processing functions may inherently interact with voiceband data traffic.
However, possible interactions are already clarified by ITU-T Recommendations and/or ETSI TISPAN specifications.
The impact of echo control (EC) and JB (jitter buffer) control is further outlined in following clauses.
6.2.1 GW behaviour (by SIP VGW, H.248 MG)
V.152 compliance guarantees correct EC and JB handling because clause 6/V.152 defines JB freezing and
clause 6.2/V.152 defines "EC and VBD mode" behaviour.
This might be an issue in non-V.152 gateways if not a similar behaviour would be supported as defined by V.152.
6.2.2 GW control of EC and JB (in SIP VGW, H.248 MG)
6.2.2.1 H.248 MG
6.2.2.1.1 Echo control
The H.248 protocol provides explicit signalling capabilities, e.g. the tdmc/ec property is used for EC control with
respect to "half-way EC" types, located at "near-end side" (i.e. PSTN domain). The EC control differs slightly between
VoIP and VBDoIP: the MGC could provide a strict EC control in case of VoIP, however, the H.248-signalled EC
settings would be overruled by V.152 behaviour in case of an autonomous transitioning to VBDoIP by the MG ("V.152
takes precedence over H.248 here"). Consequently in case of a transition back to VoIP the H.248-signalled EC settings
will be re-applied.
Correct EC behaviour might be an issue for non-V.152 gateways.
6.2.2.1.2 Jitte
...