Presentation on theme: "doc.: IEEE 802.11-03/157r0 Submission March 2003 Allen Hollister, PlantronicsSlide 1 802.11 & VoIP Allen Hollister Plantronics."— Presentation transcript:
doc.: IEEE /157r0 Submission March 2003 Allen Hollister, PlantronicsSlide & VoIP Allen Hollister Plantronics
doc.: IEEE /157r0 Submission March 2003 Allen Hollister, PlantronicsSlide 2 Router VoIP By Allen Hollister Router VoIP By Allen Hollister ter Hol lis en All Vo IP By Voice (10)RTP (12)UDP (8)IP (20)Ethernet (14)Trailer (4) Or (34) Note: To view the animation, you need PowerPoint version 2002 or later
doc.: IEEE /157r0 Submission March 2003 Allen Hollister, PlantronicsSlide 3 Router terHol lis enAll VoIP By Voice (10)RTP (12)UDP (8)IP (20)Ethernet (14)Trailer (4) Or (34) VoIP By Allen Hollister VoIP By Allen Hollister Much of QoS comes down to making the system more “TDMA” like. We want packets moving along the same path at regular intervals
doc.: IEEE /157r0 Submission March 2003 Allen Hollister, PlantronicsSlide 4 VoIP Layers VoIP sessions consist of three separate flows over the network –The outgoing media stream –The incoming media stream –The Signaling and Control stream. Streams use different routes and they use different protocols. –Media Streams use RTP/UDP because voice data is very fragile. If it doesn’t get there within 150 msec, its useless! –Signaling and Control uses TCP because it is very important that the data arrive and be accurate. The time it arrives doesn’t matter so much.
doc.: IEEE /157r0 Submission March 2003 Allen Hollister, PlantronicsSlide 5 Connection-Oriented Protocol (TCP) Destination Address Destination Address continued (address of first router) Source Address Source Address ContinuedType Total Length (16 bits) Flags Header Checksum IHLVer Fragment Offset Type of Service Source Address Destination Address (Final destination address) Options and Padding Identifier Time to LiveProtocol (6) Data Data Link Trailer Ethernet Ver.2 Header (14 bytes IP Header (20 bytes) TCP Header (min length 20 octets) Data Link Trailer Variable length Destination Port Window Acknowledgement Number Source Port Sequence Number ChecksumUrgent Pointer Options and Padding FSUPRA ReservedOffset
doc.: IEEE /157r0 Submission March 2003 Allen Hollister, PlantronicsSlide 6 TCP A port specifies an application: for example port 23 specifies Telnet, while port 25 specifies e- mail. –A list of all assigned port numbers can be found at: –A combination of a port and an IP address is a “Socket” Sequence number specifies the number assigned to the first byte of data in the current message. The first sequence number is random. The next sequence number will be the previous sequence number plus the number of octets in the last packet. It counts data octets! Acknowledgement Number: Contains the sequence number of the next byte of data the sender of the packet expects to receive Destination Port Window Acknowledgement Number Source Port Sequence Number ChecksumUrgent Pointer Options and Padding FSUPRA ReservedOffset U-Urgent Data (Urgent Pointer field becomes active) A – ACK P – Push data needs to be promptly sent to recipient R – Reset – Abort session, error S – Syn F - Finish
doc.: IEEE /157r0 Submission March 2003 Allen Hollister, PlantronicsSlide 7 Finish Acknowledge Acknowledge (4) Acknowledge (2) Acknowledge (1) Synchronize Acknowledge Disconnect Data Transfer (Slow Start) Connection TCP Startup IP Host A IP Host B Data Transfer Finish Synchronize: Sets sequence number randomly (ex 2000) Sync acknowledge from host B sets an acknowledgement number 1 greater than the original sequence number (ex 2001) then set its own random sequence number (ex 4000). Host A then send an acknowledgment number back of 1 plus the sequence number from host B (4001) Once data transmission is started, the sequence number is incremented by the number of data octets. For example if an initial sequence number is 100 and 150 octets of data are sent, then the next sequence number will be 250. Data Transfer Synchronize Acknowledge
doc.: IEEE /157r0 Submission March 2003 Allen Hollister, PlantronicsSlide 8 Complete RTP Packet Destination Address Destination Address continued Source Address Source Address ContinuedType Total Length Flags Header Checksum IHLVer Voice Data – 40 msec G.729 is 40 bytes, G bytes Data Link Trailer Fragment Offset Type of Service Source Address Destination Address Options and Padding Identifier Time to LiveProtocol(17) Destination Port Checksum Sequence Number Time Stamp Synchronization Source Contributing Source (optional) Source Port Length PTMVPCCX Ethernet Ver.2 Header (14 bytes IP Header (20 bytes) UDP Header (8 bytes) RTP Header bytes Data Link Trailer 4 bytes
doc.: IEEE /157r0 Submission March 2003 Allen Hollister, PlantronicsSlide 9 QoS in Packet Audio Delay or Latency –Problem only with two way audio, not a problem for one way (broadcast) audio such as Real Networks. –Latency in excess of 300 msec round trip (150 msec one way) is not acceptable –People perceive latency as “rudeness” and it causes people to “step” on each other words. Echo – TCLw is quite important! (TIA810a) –Echo is a problem for your listener. Typically, the person you are speaking with talks, his voice is heard in your receiver, some of that voice is then coupled to the microphone in your handset which is then transmitted back to the speaker but delayed by network latency. At certain amounts of delay and amplitude, it can make it almost impossible for the person to even speak. Any round trip delay >28 msec requires echo canceling or very good TCLw! –VoIP always has a round trip delay >28 msec! Lost Packets –More than 1% to 5% noticeable –Typical intranet <.1%, Internet <4% to 100% –Chief cause: Network congestion Jitter – Variation in Delay –Typical: Intranet < 50 msec, internet <200 msec to ∞ –Chief Cause: Queuing, CSMA systems very unpredictable Multiple paths through network causing some packets to arrive sooner than others. If VoIP phone can ’ t have predictable access to network, jitter is the result. –Adaptive jitter buffer solves jitter at expense of latency Smart jitter buffers re-adapt during silence periods. Audio quality – Vocoder issues, audio bandwidth –Various vocoder standards that allow for different audio bandwidths and different quality levels traded off against bandwidth. G Kbits/sec ADPCM 3 KHz bandwidth G Kbits/sec models voice in 10 msec chunks and sends DSP coefficients to reconstruct sound. Others: See
doc.: IEEE /157r0 Submission March 2003 Allen Hollister, PlantronicsSlide 10 QoS - Delay Delay > 150 msec one way is unacceptable –The old satellite overseas links had a one way latency of 500 ms –Typical intranet delay <50 msec, –Typical internet delay (best effort) <800 msec Chief cause of network latency: –Queuing in Routers –Going from 100 Mbit/sec to < 1 Mbit/sec Additional latency due to tradeoff of packet size vs. header overhead vs. power. –Header overhead doesn’t directly affect latency for high speed links (>1Mb/s) but does so indirectly in that more bandwidth is used. Coupled with other traffic, a congested network results that cause delay in accessing or result in lost packets. 58 bytes (464 bits) of header overhead using wired Ethernet VoIP –Robust Header Compression (RHC) can reduce this to 20 bytes 78 bytes (624 bits) using wireless Ethernet of header overhead in VoIP –When power must be saved for battery life issues, the only solution is to increase the voice sample size from, for example, 10 msec to 40 msec, then pushing this accumulated data over a high speed link in something under 5 msec, and finally putting the device into a power save mode for 35 msec. Increasing the voice sample size from 10 msec to 40 msec directly increases latency to 40 msec. You have to accumulate the entire set o samples before it can be packetized! Jitter is converted to latency through the use of a “jitter buffer”. Data is accumulated in the jitter buffer for the maximum expected arrival time for the packet. This time adds directly to latency. –If one doesn’t wait for a long enough time, packets will be lost (but latency will be less)
doc.: IEEE /157r0 Submission March 2003 Allen Hollister, PlantronicsSlide 11 Latency – Daytime, Round Trip, Scotts Valley, CA – New York Distance to NY 2582 miles. Round trip Transmission time in fiber for a 2582 mile distance is 44 ms (I assumed velocity of light is 117,000 miles/sec in fiber.) Round Trip Latency Scotts Valley to New York MaxMinAverage Daytime Nighttime
doc.: IEEE /157r0 Submission March 2003 Allen Hollister, PlantronicsSlide 12 Round Trip Latency Distance from SF to London is 6052 miles – 103 ms transmission time round trip Distance from Scotts Valley to Moscow, 7362 miles, 124 msec transmission time round trip
doc.: IEEE /157r0 Submission March 2003 Allen Hollister, PlantronicsSlide 13 Round Trip Latency - 5 mile distance
doc.: IEEE /157r0 Submission March 2003 Allen Hollister, PlantronicsSlide 14 QoS Codec Standards Codec Standards –G bits/sec (8000 samples/sec and 8 bits/sample) ADPCM with either µ law or A law applied A 20 msec audio sample requires 160 bytes –G.729 (Vocoder – Linear Predictive Coding) 8000 bits/sec Operates on 10 msec audio samples. A 20 msec audio sample would require two G.729 samples and a total of 20 bytes. –Other standards, G.729B, G.723.1, G.728, G.726, G.722 Transcoding –While it is possible to transcode between standards, it is almost never a good idea. Audio quality is lost! –Transcoding also introduces latency
doc.: IEEE /157r0 Submission March 2003 Allen Hollister, PlantronicsSlide 15 Required Data Rate
doc.: IEEE /157r0 Submission March 2003 Allen Hollister, PlantronicsSlide 16 Two VoIP Sessions Streaming Over A T1 Connection 20ms, VP 78 byte VP 20ms, VP 78 byte VP 20ms, VP 78 byte VP 20ms, VP 78 byte VP 20ms, VP 78 byte VP 20ms, VP 78 byte VP At T1 Speeds, 20 ms of G.729 (78 bytes) requires.404 ms of channel time for a single Voice Packet. At an 8 Kb/s sample rate for G.729, 20 ms of audio data results in 50 packets/sec being sent for a total of 20.2 ms of channel time. This is 2.02% of channel capacity. 20ms Router
doc.: IEEE /157r0 Submission March 2003 Allen Hollister, PlantronicsSlide Byte Data Packet Arriving At Wrong Time 20ms, VP 78 byte VP 20ms, VP 78 byte VP 20ms, VP 78 byte VP DP 512 byte DP – 2.65 ms channel time Router At T1 Speeds,.4ms of channel time is required for Voice Packets 512 byte Data Packet (requiring 2.65 ms of channel time), collides with a VP. Queuing prevents the packet from being lost, but causes a time delay or “jitter”.
doc.: IEEE /157r0 Submission March 2003 Allen Hollister, PlantronicsSlide 18 Congested Network Causing Packet Loss 20ms, VP 78 byte VP 20ms, VP 78 byte VP 20ms, VP 78 byte VP 20ms, VP 78 byte VP 20ms, VP 78 byte VP 20ms, VP 78 byte VP A VP collides with a another packet in a congested network. There is insufficient buffer space to prevent packet loss. 20ms, VP 78 byte VP 20ms, VP 78 byte VP 20ms, VP 78 byte VP DP Router
doc.: IEEE /157r0 Submission March 2003 Allen Hollister, PlantronicsSlide ms voice packet, No other traffic PBX 1ms Coder, G msec Switch routing 1 msec 2048 byte MTU data packet Wan/Lan 65 msec SF/NY 78 byte voice packet.4 msec 2048 byte MTU Data packet 11 ms T1 uplink Switch routing 1 msec Jitter buffer 40 msec Coder decompress 3 ms PBX 1 ms 78 byte voice packet.4 msec 2048 byte MTU Data packet 11 msec T1 downlink msec one way latency159.8 msec one way latency 20 ms voice packet, Data Packet ahead of VP 22 ms of the 75 ms is transmission time Estimated Latency T1 WAN uplink
doc.: IEEE /157r0 Submission March 2003 Allen Hollister, PlantronicsSlide msec one way latency No Other Traffic PBX 1 ms Coder, G msec Switch routing 1 msec 256 byte MTU data packet Wan/Lan 65 msec SF/NY 78 byte voice (20ms) packet 11 msec 256 byte MTU Data packet 37 msec 56 Kbps uplink Switch routing Jitter buffer 40 msec Coder decompress 3 ms PBX 1ms 78 byte voice (20ms)packet 11 msec 256 byte MTU Data packet 37 msec 56 Kbps downlink 233 msec one way latency – Not Good! 256 byte Data Packet Ahead Of VP 22 ms of the 75 ms is transmission time Estimated Latency 56 Kbps WAN uplink
doc.: IEEE /157r0 Submission March 2003 Allen Hollister, PlantronicsSlide 21 No Other Traffic, 11Mb/s link Switch routing 1 msec Wan/Lan 65 msec SF/NY 98 byte voice (40ms) packet.5 msec Switch routing 1msec Jitter buffer 80 msec second 802 device grabs channel with 2048 byte packet 17ms 98 byte voice (40ms)packet.5 msec T1 Uplink 1.54 Mb/s No other traffic, 1Mb/s link RF Link.1 msec 1 msec Packetization delay & G.729 coding delay 45.5 msec RF Link.1 msec 1 msec Coder Decompress 5.5 ms 2048 Data Packet, 1Mb/s link 199 msec one way latency201 msec one way latency218 msec one way latency 22 ms of the 75 ms is transmission time Estimated Latency , 40 ms VP 118 byte RF uplink packet If 2048 byte data packet remains in from of the voice packet, an additional 10ms delay occurs at the T1 uplink
doc.: IEEE /157r0 Submission March 2003 Allen Hollister, PlantronicsSlide 22 Internet QoS standards The external internet has developed a series of protocols and standards that when fully implemented, will allow telephony grade voice over the public internet. These include: –RSVP which guarantees bandwidth through all of the routers and prevents packet loss for voice packets (something else may lose packets if the network is bandwidth saturated, but it won’t be voice). –Diff-Serv uses the TOS (Type Of Service) field in the Internet Protocol (IP) to give priority over other data packets. This helps jitter and latency, two major areas of QoS concern. –MPLS (Multiprotocol Label Switching) makes a router look like a switch. The entire path for a particular voice session is reduced to the equivalent of one router “hop”. This again helps latency. –Recently the 1Gbit and now 10 Gbit Ethernet has become available. These technologies provide sufficient bandwidth as to enable very successful competition with telephony networks –These standards are relatively new and have not been fully rolled out. But when they are, then can make VoIP over the internet as good as a regular telephone connection. –These protocols are being used in private networks to provide high QoS for customers of these networks.
doc.: IEEE /157r0 Submission March 2003 Allen Hollister, PlantronicsSlide 23 RSVP Controls Queuing VP2 100 byte voice packet DP 200 byte data packet VP1 Classify VP1 Dequeue Flow Specification: Source and destination address Protocol Session Identifier (port/socket) Reserved Flow Scheduling Works in conjunction with WFQ Reserved queue for RSVP flow Queues serviced by weight VP2 Session 1, guaranteed bandwidth Session 2: Guaranteed Bandwidth DP VP2 DP (Frag) VP2VP1 DP (Frag) VP1 DP2 Discard!!
doc.: IEEE /157r0 Submission March 2003 Allen Hollister, PlantronicsSlide 24 Diff-Serv Router VP2 100 byte voice packet DP 200 byte data packet VP1 Classify VP1VP2 Dequeue DP Flow Specification: Classify based on TOS field Reserved Flow Scheduling Works in conjunction with WFQ Queues serviced by weight Highest priority Queue Second highest priority Queue DP VP2 VP1
doc.: IEEE /157r0 Submission March 2003 Allen Hollister, PlantronicsSlide 25 ToS - QoS Definitions Service TypePurposeExample 0RoutineBaggage 1PriorityStandby 2ImmediateCoach 3FlashBusiness 4Flash OverrideFirst 5CriticalPilot 6Internetwork ControlPresident Bush 7Network ControlBill Gates
doc.: IEEE /157r0 Submission March 2003 Allen Hollister, PlantronicsSlide 26 Multiprotocol Label Switching - MPLS MPLS adds a 32 bit header to the IP packet. It is used by routers to route once and switch many times –It is much faster to switch than to route. A core router may have 100,000 entries in its routing table –A predefined “tunnel” is created that each packet must follow. Each subsequent MPLS router reads the label and forwards it in “cut-through” mode. The router doesn’t have to read all of the IP headers, only those required to make the switch. –QoS is achieved by setting up high priority, high speed networks for high priority traffic and slower speed parallel networks for low or medium priority traffic. –AS far as the original packet is concerned, the routers carrying it through the MPLS network appear as a single hop.
doc.: IEEE /157r0 Submission March 2003 Allen Hollister, PlantronicsSlide 27 Signaling And Control H.323 –ITU Standard Ver 1 released Jan 1996 Ver 2 released 1998 –Consists of other standards: H.225 Call Signaling –Q.931 (ISDN Standard) –RAS (Registration, Admission, Status) H.245 Control –Originally developed to provide conferencing based communication solutions for multimedia PC’s over LAN’s –It was not designed for IP solutions. Fixed in ver 2. –What it Does Phase A – Call Setup Phase B – Initial communication and capability exchange Phase C – Establishment of audio/video communication Phase D – Call Services Phase E – Call termination SIP –IETF Standard Released July 1999 –Does the same thing as H.323 –Seen as the “up and coming” protocol. Currently H.323 is more common because it has a 3 year head start. MeGaCo (Media Gateway Control)/H.248 –Joint standard with IETF and ITU –Used to interface with the PSTN network –Works with H.323 and SIP MGCP –Predecessor to Megaco Proprietary
doc.: IEEE /157r0 Submission March 2003 Allen Hollister, PlantronicsSlide 28 SIP vs. H.323 SIPH.323 Designed by IP Networking PeopleDesigned by telecom people “ WWW mentality ” -sHTTP “ ISDN mentality ” – Q.931, ASN.1 Uses any media (but RTP is only available) H.245 (clearly defined) RTP UDP or TCP, but mostly UDPTCP Unicast or MulticastUnicast only ( not good for conferencing) ProxyGatekeeper Simpler (200 page standard)Complex (700 page standard) Extensibility (uses code numbers) Uses proprietary field in ASN.1 – Not Extensible Better suited to support intelligent user devices and better suited to implement advanced features More centrally controlled
doc.: IEEE /157r0 Submission March 2003 Allen Hollister, PlantronicsSlide 29 Objective Of H.323 Is To Enable The Exchange Of Media Streams Between H.323 Endpoints H323 Terminal H323 Gateway H323 MCU H323 Gatekeeper Packet Network Scope of H.323 PSTN
doc.: IEEE /157r0 Submission March 2003 Allen Hollister, PlantronicsSlide 30 H.323 Stack H.225 –Is itself a two part protocol. 1) A variant of ITU-T recommendation Q.931, the ISDN layer 3 specification –Used for the establishment and teardown of connections between H.323 endpoints. 2) RAS (Registration, Admission, & Status) –Used between endpoints and gatekeepers & enables the gatekeeper to manage the endpoints within its zone. H.245 is used to manage media streams between session participants. –Ensures media streams sent by one participant can be received understood by another –Operates by establishing logical channels. These channels carry the media streams between endpoints. They have a number of properties Media type Bit Rate etc H.323 Consists of a number of protocols. It specifies how these protocols will be used.
doc.: IEEE /157r0 Submission March 2003 Allen Hollister, PlantronicsSlide 31 RAS Functions Gatekeeper Discovery –Enables an endpoint to determine which gatekeeper is available to control it Registration –Endpoint registers with a particular gatekeeper. Un-registration Admission –Endpoint request to access network for purpose of participating in a session. Bandwidth is specified as part of the request Gatekeeper may turn down the request. Bandwidth change End point location –Gatekeeper translates an alias to a network address Disengage Status –Used by gatekeeper and endpoint to inform gatekeeper about call related status such as current bandwidth Resource Availability –Used from gateway to gatekeeper in order to inform gatekeeper of the gateways currently available capacity, such as bandwidth. Non-Standard –Allows proprietary information to be passed from endpoint to gatekeeper.
doc.: IEEE /157r0 Submission March 2003 Allen Hollister, PlantronicsSlide 32 Some Messages Supported By H.225 Q.931 Alerting –Sent by called endpoint to indicate that the called user is being alerted. Call Proceeding –Optional interim response sent by the called endpoint prior to sending the connect message Connect –An indication that the called user has accepted a call Setup –An initial message used to begin call establishment Setup acknowledge –Optional response to setup. –Can be forwarded from a gateway in case of PSTN interworking Release Complete –Used to bring a call to an end
doc.: IEEE /157r0 Submission March 2003 Allen Hollister, PlantronicsSlide 33 Session Initiation Protocol - SIP Performs Same Function As H.323 –WorldCom networks use SIP –ATT networks use H.323 How “net heads” want to do telephony on packet networks Very new standard –Because of head start, H.323 is more common now. –SIP is now being implemented in more new applications than H.323. Seen as “wave of the future” IETF SIP (Session Initiation Protocol) SDP (Session Description Protocol) SAP (Session Announcement Protocol) RTSP (Real Time Streaming Protocol
doc.: IEEE /157r0 Submission March 2003 Allen Hollister, PlantronicsSlide 34 SIP Addressing Addresses are known as SIP URLs (Uniform Resource Locators) –Take form of which is similar to address (but they are different) Syntax is –Sip deals with multimedia sessions, which could include voice. –SIP can interwork with traditional circuit-switched networks. SIP enables the user portion of the SIP address to be a telephone number –Such an address could be used to cause routing of media to a gateway that interfaces with the PSTN –To clearly indicate the call is a telephone number, supplement the URL with:
doc.: IEEE /157r0 Submission March 2003 Allen Hollister, PlantronicsSlide 35 SIP Network Entities SIP defines two classes of network entities –Clients An application program that sends SIP requests A client might be a PC with a headset attachment or a SIP phone. Clients can be in the same platform as a server. –SIP enable the use of proxies which can be either a client or server. –Servers An entity that responds to those requests. Four types of servers –Proxy Server –Redirect Server –User Agent Server –Registrar –SIP is a Client-server protocol.
doc.: IEEE /157r0 Submission March 2003 Allen Hollister, PlantronicsSlide 36 Servers A proxy server works like a proxy server used to access the internet from a corporate LAN –The client sends requests to the proxy and the proxy either handles the request itself or forwards it to other servers. –To those other servers, it appears as if the request is coming from the proxy rather than from the entity hidden behind the proxy. –Allows call forwarding/follow me services. Request Response Request Response Proxy
doc.: IEEE /157r0 Submission March 2003 Allen Hollister, PlantronicsSlide 37 Redirect Servers Redirect Server –Accepts SIP requests –Maps the destination address to zero or more new addresses –Returns the translated address to the originator of the request –Does not initiate any SIP requests of its own. Request Ack Request Response Redirect Server Moved, contact
doc.: IEEE /157r0 Submission March 2003 Allen Hollister, PlantronicsSlide 38 Session Description Structure Session Description Session Level Information Protocol Version Originator and Session ID Session Name Session Time Media Description1 Media Name And Transport Connection Information Media Description 2 Media Name And Transport Connection Information
doc.: IEEE /157r0 Submission March 2003 Allen Hollister, PlantronicsSlide 39 SDP Syntax Mandatory fields (test based) –v=(protocol version) Also marks the start of a session description and the end of any previous session description. –o=(session origin or creator and session identifier) –s=(session name) This field is a text string that could be displayed to potential session participants. –t=(time of session) This field provides the start time and stop time for the session. –m=(media) Indicates media type, the transport port to which the data should be sent, the transport protocol (e.g., RTP), and the media format (typically an RTP payload type. The appearance of this field also marks the boundary between session level- information and media-level information.
doc.: IEEE /157r0 Submission March 2003 Allen Hollister, PlantronicsSlide 40 SDP types - Optional i=(session information) u=(URI of description) – A web address e=( address) - of person responsible for session p=(phone number) - of person responsible for session c=(connection information) b=(bandwidth information) r=(repeat times) z=(time zone adjustment) k=(encryption key) a=(attributes)
doc.: IEEE /157r0 Submission March 2003 Allen Hollister, PlantronicsSlide 41 Subfields Some of the fields specified in SDP have sub fields. For example: –Media Information (m) has four sub fields Media type –Audio, Video, Application, Data, or Control Port –Port number to which media should be sent Transport Formats –Various types of media that can be supported –In the case of several being supported, each is listed with the preferred sent first. Examples (Format) –If a system wishes to receive voice on port and can handle speech that is coded according to G.711 mu-law, then RTP payload type 0 applies and: m=audio RTP/AVP 0 –If a system wishes to receive voice on port and can handle speech that is coded according to any G.728 (payload type 15) GSM (payload type 3) or G.711 mu-law (payload type 0) and it prefers G.728 then: m=audio RTP/AVP
doc.: IEEE /157r0 Submission March 2003 Allen Hollister, PlantronicsSlide 42 Example: Assume that Collins can support G.726, G.729, G.728, Boss can only support G.728. Note: Many SIP headers have been omitted for clarity. ACK SIP/2.0 CSeq:1 ACK Content-Length: 0 Conversation Invite SIP/2.0 Cseq: 1 INVITE Content-Length:230 Content-Typeapplication/sdp v=0 o=collins IN IP4 Station1.work.com s=vacation i=Discussion about time off work c=IN IP4 station1.work.com t=0 0 m=audio 4444 RTP/AVP a=rtpmap 2 G726-32/8000 a=rtpmap 4 G723/8000 a=rtpmap 15 G728/8000 SIP/ OK Cseq: 1 INVITE Content-Length:161 Content-Typeapplication/sdp v=0 o=manager IN IP4 Station2.work.com s=Company vacation policy c=IN IP4 station2.work.com t=0 0 m=audio 6666 RTP/AVP 15 a=rtpmap 15 G728/8000
doc.: IEEE /157r0 Submission March 2003 Allen Hollister, PlantronicsSlide 43 SIP Interworking With PSTN & H.323 PSTN Interworking –Requires a “Network Gateway” (NGW) for control interworking. NWG interfaces with SS7 database though ISUP protocol. Translates control and signaling between the two different networks. –Requires a Media Gateway to convert media between the two different networks H.323 interworking –Internet draft titled “Interworking between a SIP/SDP network and H.323” has been created. –Requires the use of a gateway that looks like a SIP user agent client or user agent server on the SIP side. On the SIP side it might include a Sip registrar function and/or proxy function. –On the H.323 side of the gateway, it could contain a gatekeeper function. –Translation between all of the H.323 messages and SIP messages occurs in this Gateway.
doc.: IEEE /157r0 Submission March 2003 Allen Hollister, PlantronicsSlide and VoIP at the Physical & Link Layers e has gone a long way to making VoIP compatible with –Many VoIP products are battery operated. Power management in e is very important –EDCF and HCF are critical for good QoS Create consistent access to the network providing low jitter, and low latency –The sidelink capability is quite helpful Can be used to save power Can be used to talk to another device (PDA remote dialer for example) while still having contact with the AP. –The PDA does the signaling and call setup, the VoIP handset handles the media. Diff-Serv at the layer 2 is a good approximation to e. –We should work with the people who are applying Diff-Serv in the routers to make sure that the standards don’t clash At the physical and link layers (given the ratification of e) we have to make sure that we don’t define something that would prohibit the use of VoIP. –Example: Someone suggested in the HTG that we could get more (payload) bits/sec by forcing a minimum size payload packet of maybe 2 to 4 KB. Doing so, would make VoIP impossible as the latency would go out of sight. The codec standard, G.729, can code 10 msec of audio in 10 bytes. Thus 2000 bytes would code 2 seconds worth of voice. The minimum latency would then be 2 sec. This is quite a bit over the 150 msec limit!!! If the committee wants to do this, they should do so with the understanding that this would be a knock-out for VoIP –Of course, for those companies that really want to do this, there will be workarounds. Get ready for a new codec standard that has 10 bytes of audio and 1990 bytes of padding Get bit rate up by creating a Robust Header Compression standard
doc.: IEEE /157r0 Submission March 2003 Allen Hollister, PlantronicsSlide 45 VoIP and At Layers Above The Link Layer Typically, 802 does not address layers above the link layer, but I understand this is more by convention than rule. If we chose to do so in this case, we would find: –Most of VoIP is already very well covered by standards at these levels. –However, the signaling standards, while complete in themselves, are not currently compatible with each other. SIP can’t talk to H.323 without a gateway! It might be fair to pick a signaling standard as “the” standard for use on products. –This would ensure that any VoIP phone could talk with any other phone, but: –It would put us in the middle of “wars” about this issue –Its unlikely that the members of this committee could agree as I suspect many are on one or the other side of this issue. –Some thought could be given to selecting a codec or set of codec’s as standard. This needs caution. All of the signaling standards have a very good mechanism for negotiating which codec to use. Selecting only one would stifle innovation and prevent growth, but: Selecting a few that all must support would ensure that communication could at least be established. –In this case, it would be very important to not prohibit other codec's from being used. –Finally, creating some good ways to do fast, seamless, handoffs as the VoIP phone wanders around a building would be a good idea. Coordinate with the IETF and other organizations who are creating a lot of new standards in this area to ensure that we remain compatible Security is likely to be a big issue for this technology. The i group has gone a long way to solving this problem, but: –It is likely in the not too distant future that complete VPN capability will become part of the operating system for portable devices. –We Should track this, and make sure that we remain compatible.
doc.: IEEE /157r0 Submission March 2003 Allen Hollister, PlantronicsSlide 46 What Can The Chip Manufacturers Do? Get the power down! Support the e standard –If you believe the PSTN system will be around for awhile, then support HCF. HCF is not necessary if you not using the PSTN system. If you are, then it makes life easier for the telephone systems people since that entire network runs from a central clock. Support the i standard.
doc.: IEEE /157r0 Submission March 2003 Allen Hollister, PlantronicsSlide 47 Thank you