| 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
This document describes a set of scenarios in which WebRTC applications have problems in Mobile Networks.
This Internet-Draft is submitted in full conformance with the provisions of BCP 78 and BCP 79.
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This Internet-Draft will expire on July 18, 2013.
Copyright (c) 2013 IETF Trust and the persons identified as the document authors. All rights reserved.
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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].
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.
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.
3GPP has standardized QoS for EPC (Enhanced Packet Core) from Release 8 [TS23.107]. 3GPP QoS policy configuration defines access agnostic 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
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 :
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.
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
The following section lists the potential problems If UE ues Public WebRTC server :
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
[I-D.wing-mmusic-ice-mobility] can be used in such scenarios to provide media stream mobility.
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
[I-D.wing-mmusic-ice-mobility] can be used in such scenarios to provide media stream 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.
This document does not define an architecture nor a protocol; as such it does not raise any security concern.
This document does not require any action from IANA.
Authors would like to thank Dan Wing, Basavraj Patil, Magnus Westerlund and Markus Isomaki for valuable inputs to the document.
Moreover WebRTC application cannot mark DSCP values on Operating Systems like Windows :
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));
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;
}