Problems with WebRTC in Mobile Networks
draft-reddy-rtcweb-mobile-00
The information below is for an old version of the document.
| Document | Type |
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| Authors | Tirumaleswar Reddy.K , John Kaippallimalil , Ram R | ||
| Last updated | 2013-01-14 | ||
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draft-reddy-rtcweb-mobile-00
RTCWEB T. Reddy
Internet-Draft Cisco
Intended status: Informational J. Kaippallimalil
Expires: July 18, 2013 Huawei
Ram Mohan. Ravindranath
Cisco
January 14, 2013
Problems with WebRTC in Mobile Networks
draft-reddy-rtcweb-mobile-00
Abstract
This document describes a set of scenarios in which WebRTC
applications have problems in Mobile Networks.
Status of this Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
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Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
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material or to cite them other than as "work in progress."
This Internet-Draft will expire on July 18, 2013.
Copyright Notice
Copyright (c) 2013 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of
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the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License.
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Notational Conventions . . . . . . . . . . . . . . . . . . . . 3
3. Scope . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3
4. Cellular Networks - QOS . . . . . . . . . . . . . . . . . . . 3
4.1. RTP Session Multiplexing . . . . . . . . . . . . . . . . . 4
4.2. Bearer Resource Modification triggered by UE . . . . . . . 5
4.3. WebRTC server deployed in 3GPP Network . . . . . . . . . . 6
4.4. Deep Packet Inspection . . . . . . . . . . . . . . . . . . 6
5. Mobility . . . . . . . . . . . . . . . . . . . . . . . . . . . 7
5.1. 3GPP SIPTO . . . . . . . . . . . . . . . . . . . . . . . . 7
5.2. IPv4 traffic offload for Proxy Mobile IPv6 . . . . . . . . 8
5.3. IPv6 Prefix with Mobility . . . . . . . . . . . . . . . . 9
6. Security Considerations . . . . . . . . . . . . . . . . . . . 9
7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 9
8. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 9
9. Informative References . . . . . . . . . . . . . . . . . . . . 9
Appendix A. OS Support . . . . . . . . . . . . . . . . . . . . . 11
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 13
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1. Introduction
The use of cellular broadband for accessing the Internet and other
data services via smartphones, tablets, and notebook/netbook
computers has increased rapidly as a result of high-speed packet data
networks such as HSPA, HSPA+, and now Long-Term Evolution (LTE) being
deployed. Browsers on these devices are becoming close to their
desktop counterparts. So, from that perspective, it is feasible to
run WebRTC applications in them. This draft enumerates problems when
WebRTC application is used on such devices in the above listed
networks.
This note focuses on QOS, traffic offload problems and does not
address other mobile network related topics like power consumption,
interface switching and congestion control related issues already
being discussed in [I-D.isomaki-rtcweb-mobile].
2. Notational Conventions
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in [RFC2119].
This note uses terminology defined in .[RFC6459]
UE, MS, MN, and Mobile : The terms UE (User Equipment), MS (Mobile
Station), MN (Mobile Node), and mobile refer to the devices that are
hosts with the ability to obtain Internet connectivity via a 3GPP
network.
3. Scope
This document can be used as a tool to design solution(s) mitigating
the encountered issues. Describing the use case allows to identify
what is common between the use cases and then would help during the
solution design phase. This note aids WebRTC server designers and
network architects to facilitate proper deployment of WebRTC in the
mobile network. WebRTC server could either be deployed in the Mobile
Network or WebRTC server deployed in a 3rd party network trusted by
Mobile Network or Public WebRTC server.
4. Cellular Networks - QOS
3GPP has standardized QoS for EPC (Enhanced Packet Core) from Release
8 [TS23.107]. 3GPP QoS policy configuration defines access agnostic
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QoS parameters that can be used to provide service differentiation in
multi vendor and operator deployments. The concept of a bearer is
used as the basic construct for which QoS treatment is applied for
uplink and downlink packet flows between the MN and gateway
[TS23.401]. A bearer may have more than one packet filter associated
and this is called a Traffic Flow Template (TFT). The IP five tuple
(IP source address, source port, IP destination, destination port,
protocol) identifies a packet filter. TFT has other attributes like
Type of Service (TOS)/DSCP. Each MN can have one or multiple bearers
associated with its registration, each supporting different QoS
characteristics. An UpLink Traffic Flow Template (UL TFT) is the set
of uplink packet filters in a TFT. A DownLink Traffic Flow Template
(DL TFT) is the set of downlink packet filters in a TFT.
The access agnostic QoS parameters associated with each bearer are
QCI (QoS Class Identifier), ARP (Allocation and Retention Priority),
MBR (Maximum Bit Rate) and optionally GBR (Guaranteed Bit Rate). QCI
is a scalar that defines packet forwarding criteria in the network.
Mapping of QCI values to DSCP is well understood and GSMA has defined
standard means of mapping between these scalars [GSMA-IR34].
Primarily LTE offers two types of bearer: Guaranteed Bit rate bearer
for real time communication, e.g., Voice calls etc. and Non-
Guaranteed bit rate, e.g., best effort traffic for web access etc.
Packets mapped to the same EPS bearer receive the same bearer level
packet forwarding treatment.
The Web Real-Time communication (WebRTC) [I-D.ietf-rtcweb-overview]
framework provides the protocol building blocks to support direct,
interactive, real-time communication using audio, video,
collaboration, games, etc., between two peers' web-browsers. WebRTC
application would use Interactive Connectivity Establishment (ICE)
protocol [RFC5245] for gathering candidates, prioritizing them,
choosing default ones, exchanging them with the remote party, pairing
them and ordering them into check lists. Once all of the above have
been completed then the participating ICE agents can begin a phase of
connectivity checks and eventually select the pair of candidates that
will be used for real-time communication.
Problems with WebRTC application in 3GPP Network are that
4.1. RTP Session Multiplexing
o [I-D.ietf-rtcweb-rtp-usage] in section 4.4 suggests to put
interactive audio, interactive video over the same 5-tuple. This
means that QoS cannot be applied to the 5-tuple, even if it were
known. QoS can only be applied by the endpoint by setting DSCP
appropriately on a per-packet basis. This problem can possibly be
solved by the proposal in [I-D.ietf-rtcweb-qos]. But 3GPP
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networks uses TFT's packet filtering information to identify and
map packets to specific bearers. So DSCP value in TFT for the
bearer will not match the DSCP value in the RTP packets sent by
the UE and 3GPP network may not honor the DSCP value in the RTP
packets. In other words both audio/video media streams would be
mapped to the same EPS bearer thus receiving the same bearer level
packet forwarding treatement.
o [I-D.ietf-rtcweb-rtp-usage] in section 4.4 also proposes not to
multiplex interactive audio, interactive video over the same
5-tuple for compatibility with legacy systems. Mobile Device
using WebRTC application in 3GPP network should prefer this method
until a technique to solve the above problem is identified and
deployed in the 3GPP network.
4.2. Bearer Resource Modification triggered by UE
If UE is using Public WebRTC server then the UE can request bearer
resource modification procedure for an E-UTRAN as explained in
section 5.4.5 of [TS23.401]. The procedure allows the UE to request
for a modification of bearer resources (e.g. Allocation or release
of resources) for one traffic flow aggregate with a specific QoS
demand. Alternatively, the procedure allows the UE to request for
the modification of the packet filters used for an active traffic
flow aggregate, without changing QoS. If accepted by the network,
the request invokes either the Dedicated Bearer Activation Procedure
or the Bearer Modification Procedure.
Problems with this approach are :
o When the UE initiates a call using WebRTC application and
candidate pairs are successfully nominated for each media stream
then WebRTC application should signal the packet filter (5-tuple),
QCI and GBR values for each media stream that would initiate
bearer resource modification or dedicated bearer activation.
WebRTC application need to be aware of the QCI/GBR values required
for the media streams.
o This means WebRTC application requires API's to indicate OS/Modem
the packet filters for each media stream to be added, modified or
deleted. WebRTC application would need API's to indicate OS/Modem
the QCI and GBR values for each of the packet filters. OS/Modem
would inturn signal this information to the 3GPP network that
would result in either bearer resource modification or creation of
Dedicated Bearer Activation Procedure.
o WebRTC application may also need API's to trigger the modification
of the GBR value for existing packet filters.
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4.3. WebRTC server deployed in 3GPP Network
Currently 3GPP networks prioritize flows by examining the SDP in SIP
signaling. As WEBRTC also uses SIP and SDP like signaling, similar
mechanisms can be used to deploy WebRTC server in 3GPP network.
o 3GPP network cannot prioritize all a=candidate lines described in
[RFC5245] until WebRTC server receives an indication of the active
media path from the controlling ICE agent. WebRTC server is aware
of the active media path only after the controlling ICE endpoint
follows the procedures in Section 11.1 of [RFC5245], specifically
to send updated offer if the candidates in the m and c lines for
the media stream (called the DEFAULT CANDIDATES) do not match
ICE's SELECTED CANDIDATES (also see Appendix B.9 of [RFC5245]).
o WebRTC server deployed in 3GPP network would need "Rx" like
interface to the Policy and Charging Rules Function (PCRF). The
PCRF is the policy server in the EPC. WebRTC server would act as
"Application Function". Dynamic PCC (policy and charging control)
rules are derived within the PCRF from information supplied by the
AF (such as requested bandwidth for the 5-tuple, Application
Identity etc). PCRF forwards PCC rules for the media stream to
the Policy Charging and enforcement function (PCEF) for Packet
scheduling, data packet (Diffserv) marking etc to allow QOS to be
provisioned in the EPC. Bearers would have be modified/created
for media streams, assigned and installed on the UE.
4.4. Deep Packet Inspection
3GPP has a current work item on "Service Awareness and Privacy
Policies" that is chartered to add DPI-related extensions to the PCC
architecture [TS23.203]. The (optional) DPI entity in the EPC is
called "Traffic Detection Function" (TDF), and it performs
application detection and reporting of detected application and its
service data flow description to the Policy Control and Charging
Rules Function (PCRF) for performing functions such as traffic
blocking, redirection, policing for selected flows.
If UE is using Public WebRTC server then
o The session signaling between the WebRTC application running in
the browser and the web server could be using TLS. Moreover
WebRTC does not enforce a particular session signaling protocol to
be used, so network gateways in 3GPP network would fail to inspect
the signalling to identify the 5-tuple used for media stream and
thus fail to prioritize media traffic. Hence derived service
identification [RFC5897] would not succeed.
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o Network Gateways by inspecting L7 traffic can only identify RTP
but fail to distinguish between IPTV vs. Multimedia, Gaming vs.
Voice Chat, Gaming vs. Voice Chat #2 etc as explained in section
3.1 - 3.3 of [RFC5897].
5. Mobility
The following section lists the potential problems If UE ues Public
WebRTC server :
5.1. 3GPP SIPTO
Given the exponential growth in the mobile data traffic, Mobile
Operators are looking for ways to offload some of the IP traffic
flows at the nearest access edge that has an Internet peering point.
This approach results in efficient usage of the mobile packet core
and helps lower the transport cost. Since Release 10, 3GPP starts
supporting of Selected IP Traffic Offload (SIPTO) function defined in
[TS23.060], [TS23.401]. The SIPTO function allows an operator to
offload certain types of traffic at a network node close to the UE's
point of attachment to the access network. Limited Mobility support
available with SIPTO is explained in section 2.3.3 of
[I-D.zuniga-dmm-gap-analysis].
If SIPTO is carried out in a Traffic offload Function (TOF) entity
located at the interface of the Radio Access Network i.e. In the
path between the Radio stations and the Mobile Gateway (MGW). The
TOF decides which traffic to offload and enforces NAT for that
traffic. The deployment of a TOF is totally transparent for the UE
and hence does not know which traffic is subject to TOF (NATed at the
TOF) and which traffic is processed by the MGW.
The problem with WebRTC application in such network is that
o TOF is not aware of the 5-tuple that will used for media and data
channels. ICE agent would gather server reflexive candidates
using STUN and relayed candidates are obtained through TURN. If
STUN messages are offloaded at TOF then UE would learn the
External IP Address/Port provided by the NAT at TOF. Similarly
ICE connectivity checks could also be offloaded at TOF. If UE
roams, though host candidate addresses may not change but NAT will
change resulting in failure to reach the remote peer for the
existing media and data channels. If the media and data channels
are offloaded at the TOF then UE Mobility would result in
disruption of media and data channel traffic.
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o If UE is using local relayed candidate to reach the remote peer
and roams out of the coverage of RNC/HNB GW then NAT between UE
and TURN server changes, so UE cannot use the previous TURN
allocations and fail to reach the remote peer using local relayed
candidate.
[I-D.wing-mmusic-ice-mobility] can be used in such scenarios to
provide media stream mobility.
5.2. IPv4 traffic offload for Proxy Mobile IPv6
Proxy Mobile IPv6 (or PMIPv6, or PMIP) is a network-based mobility
management protocol specified in [RFC5213]. Network-based mobility
management enables the same functionality as Mobile IP, without any
modifications to the host's Protocol stack. With PMIP the host can
change its point-of-attachment to the Internet without changing its
IP address.[I-D.ietf-netext-pmipv6-sipto-option] defines a way to
signal the Traffic Offload capability of a Mobile Access Gateway
(MAG) to the Local Mobility Anchor (LMA) in Proxy Mobile IP Networks.
Mobile access gateway has the ability to offload some of the IPv4
traffic flows based on the traffic selectors it receives from the
local mobility anchor. Using IP Traffic Offload Selector option
[I-D.ietf-netext-pmipv6-sipto-option] mobile access gateway will
negotiate IP Flows that can be offloaded to the local access network.
The problem with WebRTC application in such network is that
o MAG and LMA are not aware of the 5-tuple that will used for media
and data channels. If STUN messages are offloaded at local access
network then UE would learn the External IP Address/Port provided
by the NAT at local access network. Similarly ICE connectivity
checks could also be offloaded at local access network. If UE
roams out of the coverage of Local Access Network though host
candidate addresses may not change but NAT will change resulting
in failure to reach the remote peer for the existing media and
data channels. If the media and data channels are offloaded at
the local access network then UE Mobility will result in
disruption of media and data channel traffic.
o If UE is using local relayed candidate to reach the remote peer
and roams out of the coverage of Local Access Network then NAT
between UE and TURN server changes, so UE cannot use the previous
TURN allocations and fail to reach the remote peer using local
relayed candidate.
[I-D.wing-mmusic-ice-mobility] can be used in such scenarios to
provide media stream mobility.
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5.3. IPv6 Prefix with Mobility
[I-D.korhonen-dmm-prefix-properties] proposes extensions to Prefix
Information Option [RFC4861] with a mobility flag bit. This would
allow for network based mobility solutions, such as Proxy Mobile IPv6
[RFC5213] or GTP [TS.29274] to explicitly indicate that their
prefixes have mobility and therefore, the UE IP stack can make an
educated selection between prefixes that have mobility and those that
do not. WebRTC application for media streams must pick souce
addresses generated from prefixes with 'M' Flag set to 1 in Prefix
Information Option.
6. Security Considerations
This document does not define an architecture nor a protocol; as such
it does not raise any security concern.
7. IANA Considerations
This document does not require any action from IANA.
8. Acknowledgments
Authors would like to thank Dan Wing, Basavraj Patil, Magnus
Westerlund and Markus Isomaki for valuable inputs to the document.
9. Informative References
[GSMA-IR34]
"Inter-Service Provider Backbone Guidelines 5.0, 22
December 2010", September 2012.
[I-D.ietf-netext-pmipv6-sipto-option]
Gundavelli, S., Zhou, X., Korhonen, J., and R. Koodli,
"IPv4 Traffic Offload Selector Option for Proxy Mobile
IPv6", draft-ietf-netext-pmipv6-sipto-option-08 (work in
progress), January 2013.
[I-D.ietf-rtcweb-overview]
Alvestrand, H., "Overview: Real Time Protocols for Brower-
based Applications", draft-ietf-rtcweb-overview-05 (work
in progress), December 2012.
[I-D.ietf-rtcweb-qos]
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Dhesikan, S., Druta, D., Jones, P., and J. Polk, "DSCP and
other packet markings for RTCWeb QoS",
draft-ietf-rtcweb-qos-00 (work in progress), October 2012.
[I-D.ietf-rtcweb-rtp-usage]
Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time
Communication (WebRTC): Media Transport and Use of RTP",
draft-ietf-rtcweb-rtp-usage-05 (work in progress),
October 2012.
[I-D.isomaki-rtcweb-mobile]
Isomaki, M., "RTCweb Considerations for Mobile Devices",
draft-isomaki-rtcweb-mobile-00 (work in progress),
July 2012.
[I-D.korhonen-dmm-prefix-properties]
Korhonen, J., Patil, B., Gundavelli, S., Seite, P., and D.
Liu, "IPv6 Prefix Mobility Management Properties",
draft-korhonen-dmm-prefix-properties-03 (work in
progress), October 2012.
[I-D.wing-mmusic-ice-mobility]
Wing, D., Patil, P., Reddy, T., and P. Martinsen,
"Mobility with ICE (MICE)",
draft-wing-mmusic-ice-mobility-02 (work in progress),
October 2012.
[I-D.zuniga-dmm-gap-analysis]
Zuniga, J., Bernardos, C., Melia, T., and C. Perkins,
"Mobility Practices and DMM Gap Analysis",
draft-zuniga-dmm-gap-analysis-03 (work in progress),
December 2012.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC4861] Narten, T., Nordmark, E., Simpson, W., and H. Soliman,
"Neighbor Discovery for IP version 6 (IPv6)", RFC 4861,
September 2007.
[RFC5213] Gundavelli, S., Leung, K., Devarapalli, V., Chowdhury, K.,
and B. Patil, "Proxy Mobile IPv6", RFC 5213, August 2008.
[RFC5245] Rosenberg, J., "Interactive Connectivity Establishment
(ICE): A Protocol for Network Address Translator (NAT)
Traversal for Offer/Answer Protocols", RFC 5245,
April 2010.
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[RFC5389] Rosenberg, J., Mahy, R., Matthews, P., and D. Wing,
"Session Traversal Utilities for NAT (STUN)", RFC 5389,
October 2008.
[RFC5897] Rosenberg, J., "Identification of Communications Services
in the Session Initiation Protocol (SIP)", RFC 5897,
June 2010.
[RFC6459] Korhonen, J., Soininen, J., Patil, B., Savolainen, T.,
Bajko, G., and K. Iisakkila, "IPv6 in 3rd Generation
Partnership Project (3GPP) Evolved Packet System (EPS)",
RFC 6459, January 2012.
[TS.29274]
3GPP, "3GPP, "3GPP Evolved Packet System (EPS); Evolved
General Packet Radio Service (GPRS) Tunnelling Protocol
for Control plane (GTPv2-C)", 3GPP TS 29.060 8.11.0,
December 2010.", September 2012.
[TS23.060]
3GPP, ""General Packet Radio Service (GPRS); Service
description; Stage 2", June 2012.", September 2012.
[TS23.107]
3GPP, "End-to-End Quality of Service (QoS) Concept and
Architecture, Release 10, 3GPP TS 23.207, V10.0.0 (2011-
03)", September 2012.
[TS23.203]
3GPP, "3GPP, "Policy and charging control architecture",
3GPP TS 23.203 10.5.0, December 2011.", September 2012.
[TS23.401]
3GPP, "General Packet Radio Service (GPRS) enhancements
for Evolved Universal Terrestrial Radio Access Network (E-
UTRAN) access (Release 11), 3GPP TS 23.401, V11.2.0 (2012-
06).", September 2012.
Appendix A. OS Support
Moreover WebRTC application cannot mark DSCP values on Operating
Systems like Windows :
o In Windows setsockopt is completely disabled. See Knowledge Base
Article http://support.microsoft.com/kb/248611.
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o DSCP is supported and user settable on Symbian S60, Linux, MacOS
X.
The below program is to set DSCP value of 0x2E was tested on Linux successfully
(Linux k2-server-lnx1 2.6.38-8-generic #42-Ubuntu)
#include <sys/types.h>
#include <sys/socket.h>
#include <netdb.h>
#include <netinet/in.h>
#include <arpa/inet.h>
#include <stdio.h>
#include <string.h>
#include <stdlib.h>
#include <errno.h>
#include <unistd.h>
#define MSG "Hello, World!"
int
main(void) {
int sock = -1;
struct sockaddr *local_addr = NULL;
struct sockaddr_in sockin, host;
int tos = 46 << 2; /* Expedited forwarding (0x2e) */
socklen_t socksiz = 0;
char *buffer = NULL;
sock = socket(AF_INET, SOCK_DGRAM, 0);
if (sock < 0) {
fprintf(stderr,"Error: %s\n", strerror(errno));
exit(-1);
}
memset(&sockin, 0, sizeof(sockin));
sockin.sin_family = PF_INET;
sockin.sin_addr.s_addr = inet_addr("10.104.52.145");
socksiz = sizeof(sockin);
local_addr = (struct sockaddr *) &sockin;
/* Set ToS/DSCP */
if (setsockopt(sock, IPPROTO_IP, IP_TOS, &tos,
sizeof(tos)) != 0) {
printf("Error setting TOS: %s\n", strerror(errno));
}
/* Bind to a specific local address */
if (bind(sock, local_addr, socksiz) != 0) {
printf("Error binding to socket: %s\n", strerror(errno));
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close(sock); sock=-1;
exit(-1);
}
buffer = (char *) malloc(strlen(MSG) + 1);
if ( buffer == NULL ) {
printf("Error allocating memory: %s\n", strerror(errno));
close( sock ); sock=-1;
exit(-1);
}
strncpy(buffer, MSG, strlen(MSG) + 1);
memset(&host, 0, sizeof(host));
host.sin_family = PF_INET;
host.sin_addr.s_addr = inet_addr("10.106.3.95");
host.sin_port = htons(12345);
if (sendto(sock, buffer, strlen(buffer), 0,
(struct sockaddr *) &host, sizeof(host)) < 0) {
printf("Error sending message: %s\n", strerror(errno));
close(sock); sock=-1;
free(buffer); buffer=NULL;
exit(-1);
}
free(buffer); buffer=NULL;
close(sock); sock=-1;
return 0;
}
Authors' Addresses
Tirumaleswar Reddy
Cisco Systems, Inc.
Cessna Business Park, Varthur Hobli
Sarjapur Marathalli Outer Ring Road
Bangalore, Karnataka 560103
India
Email: tireddy@cisco.com
John Kaippallimalil
Huawei
5340 Legacy Drive, Suite 175
Plano Texas 75024
Email: john.kaippallimalil@huawei.com
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Ram Mohan Ravindranath
Cisco Systems, Inc.
Cessna Business Park, Varthur Hobli
Sarjapur Marathalli Outer Ring Road
Bangalore, Karnataka 560103
India
Email: rmohanr@cisco.com
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