Download Voice over Mobile IP

Experiments and Analysis of
Voice over Mobile IP
Soonuk Seol and Myungchul Kim
{suseol,mckim}@icu.ac.kr
Network Architecture Laboratory
Motivation

Voice over IP
–
–
–
–
Internet telephony is one of the most promising services
low cost, efficient bandwidth utilization, integration with data
traffic
Support only best effort service, more obstacles to deteriorate
voice quality, e.g., delay, delay jitter, packet loss, etc.
There are two competing approaches for VoIP


Mobility demand
–

ITU’s H.323 [1,2], IETF’s SIP [3]
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].
2
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
correspondent 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
applications.
Based on Mobile IP [4] for mobility support
3
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.
4
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.
(1)
CH
CH->MN
(3)
MN->CH
FA
HA
HA->FA CH->MN
(2)
CH->MN
MN
5
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
–

Redirect server
–
–
–
requests to the next hop or user-agent within an IP cloud
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.
6
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

210.107.132.3
210.107.131.181
CH
HA
210.107.132.83
na-ep1.icu.ac.kr
210.107.132.0 net
gateway
210.107.131.0 net
210.107.132.81
210.107.143.209
Router2
na-router2.icu.ac.kr
210.107.143.208 net
210.107.143.214
210.107.143.210
FA1
i3ebs1.icu.ac.kr
FA2
i3ebs2.icu.ac.kr
210.107.143.221
210.107.143.217
210.107.143.216 net
MH
210.107.143.212 net
210.107.143.220 net
210.107.132. 66
IEEE 802.11 PC Card 11 Mbit/s
7
RTP packet format
Version
Length
Type of service
Identification
Time to live (TTL)
Total length (in byte)
Flags
Protocol
Fragment offset
Header checksum
Source IP address
Destination IP address
Option (if any)
Ver
Source port
Destination port
Datagram length
Checksum
Payload type
Sequence number
Timestamp
Synchronization source identifier
Application data

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)
8
Encapsulation delay

Encapsulation and decapsulation delay in Mobile IP: ~ 1ms
210.107.132.83
na-ep1.icu.ac.kr
HA
210.107.131.181
gateway
CH
210.107.132.3
HA
210.107.132.0 net
FA1
210.107.131.0 net
210.107.132.81
210.107.143.209
CH
Router2
na-router2.icu.ac.kr
210.107.143.208 net
210.107.143.212 net
210.107.143.210
FA1
i3ebs1.icu.ac.kr
210.107.143.217
210.107.143.216 net
210.107.132. 66
–
210.107.143.21
FA2
4
i3ebs2.icu.ac.kr
x
210.107.143.221
210.107.143.220 net
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
HA
FA1
CH
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.
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Interarrival time in one-way calls
over conventional IP
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–
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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)
in te ra rriva l tim e (s e c )
0.03500
0.03000
0.02500
0.02000
0.01500
0.01000
0.00500
0.00000
m a x = 0.02391
m in = 0.01611
1000
0
100
200
300
400
500
Frequency
–
100
600
700
10
1
0
10
20
30
40
interarrival time (ms ec)
10
Interarrival time in one-way calls
over Mobile IP
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–
–
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)
in te ra rriva l tim e (s e c )
0.03500
0.03000
0.02500
0.02000
0.01500
0.01000
0.00500
0.00000
m a x = 0.03108
m in = 0.00901
1000
0
100
200
300
400
500
Frequency
–
600
700
100
10
1
0
10
20
30
40
interarrival time (ms ec)
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Interarrival time in voice conversation(1)


Bi-directional voice conversation for 60 seconds.
Average: 20ms, overall within 42ms for every case:
(a) IP
(c) Mobile IP with 5
times of handoffs
(b) Mobile IP without handoffs
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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.
c->m
CH
HA
h->FA1 c->m
FA1
FA2
MH
h->FA2 c->m
CH
HA
FA1
FA2
MH
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Interarrival time under background traffic
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five extra sessions for MN with different hosts, totally 6300 packets
(~2min) for each call
Packet losses:
5 for normal IP
6 for mobile IP
98%
interarrival time (msec)
normal IP
packets
Mobile IP packets
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The longest : Normal IP = 25 ms, Mobile IP = 30 ms
98% of packets = 18 ~ 22 ms
Traditional packet loss
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Bandwidth of GSM codec

SIP application, Linphone with GSM
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One frame of 33 bytes in a single packet
Size of packet headers : 54 bytes
Frame duration : 20 ms
frames/pkt
1
2
3
4
5
6
7
8
9
10
11
12
pkts/sec
payload
50.00
25.00
16.67
12.50
10.00
8.33
7.14
6.25
5.56
5.00
4.55
4.17
bytes
33
66
99
132
165
198
231
264
297
330
363
396
bits/sec
pkt size
13200
13200
13200
13200
13200
13200
13200
13200
13200
13200
13200
13200
bytes
87
120
153
186
219
252
285
318
351
394
417
450
bits/sec
34800
24000
20400
18600
17520
16800
16286
15900
15600
15360
15164
15000
% optimal latency
264 %
182 %
155 %
141 %
133 %
127 %
123 %
120 %
118 %
116 %
115 %
114 %
ms
20
40
60
80
100
120
140
160
180
200
220
240
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Total Data Size for Different frames/pkt
One-way voice data
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Totally, 297 Kbytes for 180 sec (one frame : 33 bytes
IP & UDP headers: add 54 bytes
Encapsulation (from HA to FA): adds 20 bytes
one frame per packet (f/p = 1)
CH
FA
HA
basic headers
(54 bytes * 3)
.
–
1000
K b yte, 10 packets

800
)
three frames per packet (f/p = 3)
MH
CH
FA
HA
MH
voice data
header for tunneling
headers for tunneling
(33-byte
frame
*
3)
(20 bytes)
(20 bytes * 3)
voice data
basic header
(33-byte frame * 3)
(54 bytes)
headers fo r tunneling betw een H A and FA
basic headers
vo ice data
the num ber o f packets
600
400
200
0
1
2
3
4
5
6
7
8
9
10
11
12
fram es / packet
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The Desirable Number of Frames

Mobile IP Network
–
need to save the bandwidth (esp., wireless network)
lower bound of 99% confidence interval
100%
120
80%
The maximum packet
transmission delay
permitted (ms)
150
90
60%
88.9 ms
60
46.1%
40%
30
20%
0
0%
-30
1
2
3
4
5
6
7
-60
-90

bandwidth save
8
9
10
11
12
-20%
-40%
frames / packet
-60%
bandwidth save
End to end delays
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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
(150-60) are remained for packet transfer.
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Conclusion and Future work


Feasibility of Mobile IP-based SIP
–
Mobile IP’s encapsulation and decapsulation delay is short
enough for interactive audio applications.
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Interarrival time does not vary much.
Desirable number of frames per packet
–

Sends three frames per packet to reduce loads on the campussized network.
Future work
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Simulate SIP over Mobile IP for large scaled networks
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study various kinds of codecs in the same context and in terms
of the number of hops.
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delay-aware and/or load-aware scheme for Internet Telephony
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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,”
Eighth International Conference on Computer Communications and Networks, 1999.
[6] Elin Wedlund and Henning Schulzrinne, “Mobility Support using SIP,” Proceedings of the second ACM
International Workshop on Wireless Mobile Multimedia (WoWMoM), 1999.
[7] X. Zhao, C. Castelluccia, and M. Baker, “Flexible Network Support for Mobility,” in Proceedings of
Mobicom, 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
Annual 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.
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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-mobileipoptim-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.
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