Network Architecture Laboratory

Experiments and Analysis of Voice over Mobile IP

Soonuk Seol and Myungchul Kim

{suseol,mckim}@icu.ac.kr

2

Motivation

Voice over IP – Internet telephony is one of the most promising services– low cost, efficient bandwidth utilization, integration with data traffi

c– Support only best effort service, more obstacles to deteriorate vo

ice quality, e.g., delay, delay jitter, packet loss, etc.– There are two competing approaches for VoIP

ITU’s H.323 [1,2], IETF’s SIP [3]

Mobility demand– VoIP needs to support most functionalities that the current PSTN

does, especially mobility support. All-IP trends

– Recently, it is believed all mobility-related functionality should be handled at the IP (network) layer [10,11,12,13].

3

Related Work

Extensions to H.323 for mobility [8,9] :– Additional messages and functionalities to H.323 system– Requires applications to perform mobility management

Mobility support to SIP– Moh et al. [5]

Address several major issues for supporting mobility on SIP– Wedlund and Schulzrinne [6]

An application level approach for real-time mobile communication. Mobility support is limited to SIP-aware applications and SIP-aware correspond

ent hosts. Networks should support DHCP to assign IP addresses. Overhead with mobile IP

– A waste of resources to keep duplicated information about the hosts current address. (both in SIP servers and Home agents)

In our experiments– Need a homogeneous mobility solution regardless of wireless interfaces and applic

ations. – Based on Mobile IP [4] for mobility support

4

What we have achieved

Examine the feasibility of Voice over Mobile IP for Internet telephony

– Investigate various factors that affect delay, packet loss, and load on the network

– Experiment with encapsulation and decapsulation delay time and interarrival time in many aspects, comparing with normal IP.

Find the desirable number of frames per packet in Mobile IP as a function of packet transmission delay and bandwidth utilization.

5

Backgrounds

Mobile IP– Allows a mobile node to communicate with other nodes

transparently in spite of address change due to its mobility

– Triangular routing problem which increases delays– Route optimization solves the problem by using binding

updates.

FA HA

CH

HA- >FA CH- >MN

CH- >MN

CH- >MN

MN- >CH

MN

(1)

(2)

(3)

6

Backgrounds

Session Initiation Protocol (SIP)– SIP allows two or more participants to establish a session

consisting of multiple media streams.– In SIP, callers and callees are identified by SIP address. – When making a SIP call, a caller first locates the

appropriate server and then sends a SIP request.– SIP server can act in two different modes

Proxy server – requests to the next hop or user-agent within an IP cloud

Redirect server– informs their clients of the address of the requested server– allow for the client to contact that server directly

– In our experiment, we make calls through peer-to-peer communications without any server.

7

Testbed Configuration

Mobile IP: Dynamics, http://www.cs.hut.fi/Research/Dynamics/ SIP: Linphone, http://www.linphone.org, GSM codec is used Analysis with TCPDUMP (for capturing packets) and Ping

Router2

FA1

210.107.132.81 210.107.143.209

210.107.143.210

210.107.143.217

na-router2.icu.ac.kr

HA

210.107.132.83

na-ep1.icu.ac.kr

i3ebs1.icu.ac.kr

210.107.132.3

MH 210.107.132. 66

CH

210.107.131.181

gateway

210.107.131.0 net 210.107.132.0 net

210.107.143.208 net

FA2

210.107.143.214

210.107.143.221

i3ebs2.icu.ac.kr

210.107.143.212 net

210.107.143.216 net 210.107.143.220 net

IEEE 802.11 PC Card 11 Mbit/s

8

RTP packet format

Length of a packet : 87 bytes– IP header : 20 bytes, IP option : 14 bytes– UDP header : 8 bytes– RTP message : 45 bytes ( RTP header : 8bytes, Voice data: 33 bytes)

Version Length Type of service Total length (in byte)

Identification Fragment offset

Time to live (TTL) Header checksum

Source IP address

Destination IP address

Source port Destination port

Datagram length Checksum

Ver Payload type Sequence number

Protocol

Flags

Timestamp

Synchronization source identifier

Application data

Option (if any)

9

Encapsulation delay

Encapsulation and decapsulation delay in Mobile IP: ~ 1ms

– Measure the encapsulation and decapsulation delay by configuring the routing path between MH and CH in mobile IP to be identical to that of not using mobile IP.

Router2

FA1

210.107.132.81

210.107.143.209

210.107.143.210

210.107.143.217

na-router2.icu.ac.kr

HA

210.107.132.83

na-ep1.icu.ac.kr

i3ebs1.icu.ac.kr

210.107.132.3

210.107.132. 66

CH

210.107.131.181 gateway

210.107.131.0 net

210.107.132.0 net

210.107.143.208 net

FA2

210.107.143.21

4

210.107.143.221

i3ebs2.icu.ac.kr

210.107.143.212 net

210.107.143.216 net 210.107.143.220 net

x

y (mobile IP pkt) y’ (normal IP pkt)

2x = 3.2 ms, x=1.6 ms x+y = 4.2 ms y = 2.6 ms Assuming x=y’, y-y’ = y-x = 1ms

FA1 HA

CH

FA1 HA

CH

10

Interarrival time in one-way calls over conventional IP

– Sending interval : 20 ms– Interarrival time : 19.95 ~ 20.05 ms with 99%

confidence– Standard deviation : 0.5 ms– Number of samples : 700 packets (14 seconds)

interarrival time (sec)

max = 0.02391

min = 0.01611

0.000000.005000.010000.01500

0.020000.025000.030000.03500

0 100 200 300 400 500 600 700

1

10

100

1000

0 10 20 30 40

interarrival time (msec)

Fre

que

ncy

11

Interarrival time in one-way calls over Mobile IP

– Sending interval : 20 ms– Interarrival time : 19.91 ~ 20.09 ms with 99%

confidence– Standard deviation : 0.89 ms– Number of samples : 700 packets (14 seconds)

interarrival time (sec)

max = 0.03108

min = 0.00901

0.00000

0.005000.01000

0.01500

0.02000

0.025000.03000

0.03500

0 100 200 300 400 500 600 700

1

10

100

1000

0 10 20 30 40

interarrival time (msec)F

req

uenc

y

12

Interarrival time in voice conversation(1)

Bi-directional voice conversation for 60 seconds. Average: 20ms, overall within 42ms for every case:

(a) IP

(b) Mobile IP without handoffs

(c) Mobile IP with 5 times of handoffs

13

Interarrival time in voice conversation(2)

Overall packets arrive within 42 ms. (be made up with buffers) No many differences during the handoff time.

The reason is that a mobile node – can receive packets from the old foreign agent.– gets a care-of address from the FA not from the DHCP server.– Cells are overlapped enough.

FA1

HA CH

FA2

h->FA1c->m

c->m

h->FA2c->m

MH

FA1

HA CH

FA2

MH

14

Interarrival time under background traffic

– five extra sessions for MN with different hosts, totally 6300 packets (~2min) for each call

– The longest : Normal IP = 25 ms, Mobile IP = 30 ms– 98% of packets = 18 ~ 22 ms– Traditional packet loss

normal IP packets

Mobile IP packets

Packet losses: 5 for normal IP 6 for mobile IP

interarrival time (msec) 98%

15

Bandwidth of GSM codec

SIP application, Linphone with GSM– One frame of 33 bytes in a single packet– Size of packet headers : 54 bytes– Frame duration : 20 ms

frames/pkt

pkts/sec payload bits/sec

pkt size bits/sec

% optimal

latency bytes bytes ms

1 50.00 33 13200 87 34800 264 % 202 25.00 66 13200 120 24000 182 % 403 16.67 99 13200 153 20400 155 % 604 12.50 132 13200 186 18600 141 % 805 10.00 165 13200 219 17520 133 % 1006 8.33 198 13200 252 16800 127 % 1207 7.14 231 13200 285 16286 123 % 1408 6.25 264 13200 318 15900 120 % 1609 5.56 297 13200 351 15600 118 % 180

10 5.00 330 13200 394 15360 116 % 20011 4.55 363 13200 417 15164 115 % 22012 4.17 396 13200 450 15000 114 % 240

16

Total Data Size for Different frames/pkt

One-way voice data– Totally, 297 Kbytes for 180 sec (one frame : 33 bytes )– IP & UDP headers: add 54 bytes – Encapsulation (from HA to FA): adds 20 bytes

0

200

400

600

800

1000

1 2 3 4 5 6 7 8 9 10 11 12frames / packet

Kbyt

e, 1

0 pa

cket

s

. headers for tunneling between HA and FAbasic headersvoice datathe number of packets

FA HA CH MH FA HA CH MH

one frame per packet (f/p = 1) three frames per packet (f/p = 3)

basic headers(54 bytes * 3)

voice data(33- byte frame * 3)

basic header(54 bytes)

header for tunneling(20 bytes)

headers for tunneling (20 bytes * 3)

voice data(33- byte frame * 3)

17

The Desirable Number of Frames

Mobile IP Network– need to save the bandwidth (esp., wireless network)

End to end delays– Smaller than 150 ms : not perceived (our goal)– Between 150 and 400 ms : acceptable but not ideal– If f/p=3: about 60ms’ latency to aggregate three frames. The rest 90ms (15

0-60) are remained for packet transfer.

88.9 ms

46.1%

-90

-60

-30

0

30

60

90

120

150

1 2 3 4 5 6 7 8 9 10 11 12

frames / packet

The

ma

xim

um p

ac

ket

tran

sm

iss

ion

de

lay

per

mitt

ed

(ms)

-60%

-40%

-20%

0%

20%

40%

60%

80%

100%

bandwidth save

lower bound of 99% confidence interval bandwidth save

18

Conclusion and Future work

Feasibility of Mobile IP-based SIP– Mobile IP’s encapsulation and decapsulation delay is short enough for in

teractive audio applications.– Interarrival time does not vary much.

Desirable number of frames per packet– Sends three frames per packet to reduce loads on the campus-sized net

work.

Future work– Simulate SIP over Mobile IP for large scaled networks– study various kinds of codecs in the same context and in terms of the nu

mber of hops. – delay-aware and/or load-aware scheme for Internet Telephony

19

References

[1]   Gary A. Thom, “H.323: the Multimedia Communications Standard for Local Area Networks,” IEEE Communications Magazine, December 1996.

[2]   ITU-T Rec. H.323v2, “Packet Based Multimedia Communications Systems,” March 1997.[3]   M. Handley et al., “SIP: Session Initiation Protocol,” IETF RFC 2543, March 1999.[4]   C. Perkins, “IP Mobility Support,” RFC 2002, IETF, October 1996.[5]   Melody Moh, Gregorie Berquin, and Yanjun Chen, “Mobile IP Telephony: Mobility Support of SIP,” Ei

ghth International Conference on Computer Communications and Networks, 1999.[6]   Elin Wedlund and Henning Schulzrinne, “Mobility Support using SIP,” Proceedings of the second AC

M International Workshop on Wireless Mobile Multimedia (WoWMoM), 1999.[7]   X. Zhao, C. Castelluccia, and M. Baker, “Flexible Network Support for Mobility,” in Proceedings of Mo

bicom, October 1998.[8] ITU-T Draft Recommendation H.MMS.1, “Mobility for H.323 Multimedia Systems,” March 2001. [9] Wanjiun Liao, “Mobile Internet Telephony: Mobile Extensions to H.323,” INFOCOM ’99. Eighteenth A

nnual Joint Conference of the IEEE Computer and Communications Societies. Proceedings. IEEE, June 1999.

[10] Ramachandran Ramjee, Thomas F. La Porta, Luca Salagrelli, Sandra Thuel, and Kannan Varadhan, “IP-based Access Network Infrastructure for Next-Generation Wireless Data Networks,” IEEE Personal Communications, August 2000.

[11] Shingo Ohmori, Yasushi Yamao, and Nobuo Nakajima, “The Future Generations of Mobile Communications Based on Broadband Access Technologies,” IEEE Communications Magazine, December 2000.

20

References (cont.)

[12] Ramón Cáceres and Venkata N. Padmanabhan, “Fast and Scalable Wireless Handoffs in Supports of Mobile Internet Audio,” Mobile Networks and Applications 3, December 1998.

[13] Mihailovic, A., Shabeer, M., and Aghvami, A.H., “Multicast for Mobility Protocol (MMP) for Emerging Internet Networks,” The 11th IEEE International Symposium on Personal, Indoor and Mobile Radio Communications (PIMRC), 2000.

[14] H. Schulzrinne and J. Rosenberg, “A Comparison of SIP and H.323 for Internet Telephony,” http://www.cs.columbia.edu/~hgs/sip/papers.html.

[15] James F. Kurose and Keith W. Ross, “Computer Networking – A Top-Down Approach Featuring the Internet”, Addison Wesley Longman, 2001.

[16] Charles Perkins and David B. Johnson, “Route Optimization in Mobile IP,” draft-ietf-mobileip-optim-11.txt (Work in progress), September 2001.

[17] David B. Johnson and Charles Perkins, “Mobility Support in IPv6,” draft-ietf-mobileip-ipv6-13.txt (Work in progress), July 2001.

[18] Dynamics – HUT Mobile IP, available at http://www.cs.hut.fi/Research/Dynamics/index.html.

[19] Linphone – a SIP application, available at http://simon.morlat.free.fr/english/linphone.html.