1 / 22

SIP in 3G

SIP in 3G. HUT S-38.130 Spring 2001 Tuomo Sipilä Nokia Research Center. SIP in 3G: Content. Background 3GPP R5 architecture Packet Core Network IP Multimedia Subsystem Requreiments Architecture SIP protocol in 3G 3G SIP requirements Problems Conclusions. Background.

aderyn
Download Presentation

SIP in 3G

An Image/Link below is provided (as is) to download presentation Download Policy: Content on the Website is provided to you AS IS for your information and personal use and may not be sold / licensed / shared on other websites without getting consent from its author. Content is provided to you AS IS for your information and personal use only. Download presentation by click this link. While downloading, if for some reason you are not able to download a presentation, the publisher may have deleted the file from their server. During download, if you can't get a presentation, the file might be deleted by the publisher.

E N D

Presentation Transcript


  1. SIP in 3G HUT S-38.130 Spring 2001 Tuomo Sipilä Nokia Research Center

  2. SIP in 3G: Content • Background • 3GPP R5 architecture • Packet Core Network • IP Multimedia Subsystem • Requreiments • Architecture • SIP protocol in 3G • 3G SIP requirements • Problems • Conclusions

  3. Background • 3G is known as UMTS in Europe, as IMT-2000 in Japan • The standarisation work for IP based multimedia started in Autumn 1999 based on input from 3G.IP • Targets to standardise the required enhancements for the 3G network so that • IP telephony and multimedia can be provided with equal user perceived quality as with the current mobile network services • 3G network can function fully based on packet and IP connections (without traditional circuit switched domain) • IP multimedia would in the future provide via IP a wider and more flexible service set than the current networks • SIP was selected as the signalling protocol for IP Multimedia in Spring 2000

  4. 3GPP Rel5 system architecture • Radio Access Network Domain (RAN) For radio access WCDMA based UTRAN or GSM/EDGE based GERAN • Circuit Switched Core Network Domain (CS CN) for Circuit switched services • Packet Switched Core Network Subsystem (PS CN) for provision of PS connectivity services • IP Multimedia Core Network Subsystem (IMSS) for the IP base multimedia services, IPv6 based system ! • Service Subsystem for operator specific services (e.g. IN and OSA) • Subsystem independent evolution and access independency is the principle NOTE: Not all interfaces are shown !

  5. Internet Services Subsystem RNC BSC IP PS CN Architecture Key issues • Normally Terminal activated the PDP contexts (between GGSN and UE) • Four QoS classes defined for packet connection • Primary PDP context activation: issue IP address to the terminal • Secondary PDP context: new flow with new QoS with same IP address • Traffic Flow Template: Filters the IP flows to the right PDP context • Gi and Go interface towards IP Multimedia Subsystem (Go for policy control) HSS UTRAN CAP over SS7/IP Gr Gc Iu Gi/Go Gn Gn SGSN GGSN GERAN Iu PS CN domain

  6. 3G QoS Classes in Packet Core network

  7. PDP Context activation

  8. External IP networks and other IMS networks Applications & Services Legacy mobile signalling network R-SGW SCP P/I/S-CSCF Sc Ms PSTN/ Legacy /External Mh MRF BGCF Mc Mm S-CSCF Mm Cx HSS Mw Cx Gc I-CSCF Mi Gi Mk Mw BGCF Mg Go Mw GGSN P-CSCF Mj T-SGW MGCF Gi Mc MGW IM SS architecture • Gi interface from GGSN to external networks is not shown in the figure

  9. Requirements for IMSS • at least equal end-to-end QoS for voice as in circuit switched (AMR Codec based) wireless systems • equal privacy, security or authentication as in GPRS and circuit switched services • QoS negotiation possibility for IP sessions and media components by both ends • access independence i.e. the IP Multimedia network and protocols evolve independently of radio access (WCDMA, EDGE/GSM/GPRS, WLAN etc) • applications shall not be standardised • IP policy control possible i.e the operators shall have the means to control which IP flows use the real-time QoS bearers • automated roaming with the services in home and visited network • hide the operator network topology from users and home/visited network • the resources shall be made available before the destination alerts • adressing with SIP URL or E.164 number • procedures for incoming and outgoing calls, emergency calls, presentation of originator identity, negotiation, accepting or rejecting incoming sessions., suspending, resuming or modifying the sessions • user shall have the choice to select which session components reject or accept

  10. Network Elements (1/3) • HSS (Home Subscriber Server) • User Identification, Numbering and addressing information. • User Security information: Network access control information for authentication and authorization • User Location information at inter-system level; HSS handles the user registration, and stores inter-system location information, etc. • The User profile (services, service specific information…) • P-CSCF (Proxy Call State Control Function) • First contact point for UE within IM CN subsystem forwards messages to S-CSCF • Is like proxy or user agent in RFC 2543 (SIP) • Is discovered using DHCP during registration or the address is sent with PDP context activation • May perform number analysis (e.g., detect local service numbers) • Detect and forward emergency calls • Call monitoring and logging (e.g., billing verification) • Authorization of resource usage

  11. Network Elements (2/3) • S-CSCF (Serving Call State Control Function) • Maintains call state required to provide call related services • Interacts with Services Subsystem • Controls MRF • Monitors sessions for billing purposes • I-CSCF (Interrogating Call State Control Function) • "is the contact point within an operator's network for all connections destined to a subscriber of that network operator, or a roaming subscriber currently located within that network operator's service area" • can be reagarded as a firewall • Routes SIP requests from another networks to S-CSCF and MGCF • May hide service provider's network topology • Selects S-CSCF during registration

  12. Network Elements (3/3) • MGCF (Media Gateway Control Function) • Protocol conversion between ISUP and SIP • Routes incoming calls to appropriate CSCF • Controls MGW resources • MGW (Media Gateway) • Transcoding between PSTN and 3G voice codecs • Termination of SCN bearer channels • Termination of RTP streams • T-SGW (Transport Signalling Gateway) • Maps call related signalling from/to PSTN/PLMN on an IP bearer • Provides PSTN/PLMN <-> IP transport level address mapping • MRF (Multimedia Resource Function) • Performs multiparty call and multi media conferencing functions • BGCF (Breakout Gateway control function ) • selects the network in which the PSTN interworking should occur • selects the MGCF which will perform the interworking

  13. Roaming model User A A’s visited network Required onregistration,optional on sessiion establish A’s home network S-CSCF P-CSCF I-CSCF I-CSCF Optional User B I-CSCF I-CSCF P-CSCF Required onregistration,optional on sessiion establish S-CSCF B’s visited network B’s home network - P-CSCF - Proxy CSCF (Call Session Control Function). The terminals point of contact in the visited network after registration. - I-CSCF - Interrogating-CSCF. Responsible for finding the S-CSCF at registration. May also perform hiding of the S-CSCF network architecture. - S-CSCF - Serving-CSCF. Responsible for identifying user’s service priveleges. Responsible for selecting access to home network application server (service platform) and for providing access to that server

  14. SIP in IMSS interface Gm: P-CSCF - UE Mw: P-CSCF - S-CSCF and P-CSCF - I-CSCF Mm: S/I-CSCF - external IP networks & other IMS networks Mg: S-CSCF - BCGF Mk: BCGF - external IP networks & other IMS networks SIP+ is used to interface the Application servers: S-CSCF- SIP Application server S-CSCF- Camel Server S-CSCF-OSA Service Server SLF HSS AS Cx Cx Dx Gm Mw Mw UA P-CSCF I-CSCF S-CSCF Mg Mg BCGF MGCF Mc MGW SIP in interfaces

  15. Current 3GPP SIP procedures • Local P-CSCF discovery • Either using DHCP or carrying address in the PDP context • S-CSCF assignment and cancel • S-CSCF registration • S-CSCF re-registration • S-CSCF de-registration (UE or network initiated) • Call establishment procedures separated for • Mobile origination; roaming, home and PSTN • Mobile termination; roaming, home and PSTN • S-CSCF/MGCF - S-CSCF/MGCF; between and within operators, PSTN in the same and different network • Routing information interrogation • Session release, Session hold and resume • Anonymous session establishment • Codec and media flow negotiation (Initial and changes) • Called ID procedures • Session redirect, Session Transfer

  16. Some requirement solutions Key issues: A) Mobile terminated calls • 1) have network initiated PDP Context activation (required static IP address) • discussion ongoing on push services • options 1: a new element to link the IMSI with the dynamic IP address allocation • option 2: use SMS to trigger PDP activation in the terminal • 2) provide an always on PDP context (signalling PDP context) • the P-CSCF address to the terminal • either during the PDP context activation or • after PDP activation with DHCP procedures, then with DNS to find the IP address • both options possible with current specs B)avoid alerting before the resources are available • 2 phase call setup C) Should SIP use a signalling channel on Radio interface ? • If yes the capabilities needs to be limited and message compression used • will limite the usage of SIP to signalling protocol only

  17. Registeration

  18. Mobile initiated call setup 1-22: Session description exchange 23-31: Resource reservation 32- 43: Alerting 44-52: Answering the call

  19. Example of INVITE message

  20. SIP protocol requirements in 3GPP • addition of routing PATH header to the SIP messages to record the signalling path from P-CSCF to S-CSCF • location information in the INVITE message to carry the location of the terminal (for instance Cell ID) • emergency call type is needed to indicate the type of emergency call i.e. is it police, ambulance etc. • filtering of routing information in the IM SS before the SIP message is sent to the terminal to hide the network topology from terminal • refresh mechanism inside IM SS • Network-initiated de-registration • 183 Session Progress provisional response for INVITE to ensure that the altering is not generated before PDP contexts for session are activated • Reliability of provisional responses - PRACK method to acknowledge the 183 message • Usage of session timers to keep the SIP session alive • Indication of resource reservation status - COMET method • Security for privacy • Extensions for caller preferences and callee capabilities • Media authorisation token for the Policy Control function to authorise the PDP context with SIP connection in the UE

  21. Problems • architecture complexity • call establishment delay problems due to the signalling taking place on multiple levels (RAN, PS CN, IMSS). • establishing a call there will be 6 round trip times (RTT) end to end on SIP level + PDP context reservations • guarantees of QoS • Several elements and several IP based interfaces • lengthy standardisation time • suitability of the SIP protocol for the radio interface, long character based messages, compression needed • IETF and 3GPP standardisation co-operation • Terminal complexity

  22. Conclusions: 3GPP specifics for SIP • the architecture of the IMSS is defined based on 3G model (home and visited), messages run always via S-CSCF • Registration is mandatory • The CSCFs interrogate the SIP and SDP flows either actively modifying the messages or reading the data, also the I-CSCF hides the names of CSCF behind it • Codec negotiations in 3GPP do not allow different codecs in different directions • in 3G networks there is a separation of UNI and NNI interface • due to radio and packet core functionality there are some change proposals to the SIP and SDP • due to the P-CSCF - S-CSCF interface and the 3G roaming mode there are some requirements to the SIP and SDP protocols • in 3G SIP is used also to interface the application development elements, they set requirements for SIP and SDP protocols THUS • SIP is suitable for 3G if the problems (call delays, SIP length, QoS) can be solved • Specification work shall take still some time • 3G and SIP should provide enhaced and rich services NOT be ONLY the replacement for CS

More Related