Network Applications in the Voice-Over-Net Age Jongwon Yoon, Sriram Subramanian University of Wisconsin-Madison {yoonj, srirams}@cs.wisc.edu ABSTRACT vice provider [4] ranking them ahead of other services like VoIP provides home users with a low cost alternative to tra- Comcast Digital Voice, ATT Call Vantage, Time Warner Ca- ditional telephone services. In a typical deployment of a ble etc. The study measured Audio Performance metrics VoIP service within a home, various network applications (MOS (Mean Opinion Score) and Audio Delays, Signal Lev- coexist with VoIP call traffic. In this empirical study, we use els, Silence Levels and Codec used), Network Metrics (Packet the Vonage VoIP service as a case-study to examine the po- Counts, Latency, Ordering and Jitter). tential impact of traditional network-based applications on the performance of VoIP calls, and vice-versa. 2. VONAGE MECHANISM Applications tend to use different transport protocols such Vonage is a publicly held commercial VoIP network and as TCP and UDP with varied packet sizes and data rates. SIP company that provides telephone service via a broad- We identify the impact of different characteristics of the ap- band connection. Vonage phone is a replacement of the plication flows on Vonage calls in terms of delay, jitter and regular phone with low cost and Vonage service doesn’t re- packet loss. We find that smaller packets tend to get prefer- quire any softphone. Vonage users use traditional analog ential treatment at Vonage routers, and this would affect the phones and only needs an internet connection for VoIP ser- quality of the VoIP traffic drastically causing jitter, loss and vice. Therefore users can continue their Vonage service any- throughput variations. where as long as they have an Internet connection and von- While our observations are specific to Vonage, we believe age equipments. that the broad conclusions can be applied to guide the design of future VoIP software and hardware. I N T E R N E T S L M o d e m 1. INTRODUCTION D Current networks were originally designed for the trans- mission of data, and are inadequate for the transmission of o l l e c t H U B P C ( C k e t T r c e ) real-time and multimedia data since packet loss and delays c a a significantly degrade the quality. The routing policies could P also result in packets arriving out of order, which could have c t significant impact on the real-time application traffic, such e R O U T E R P C ( I n j B c k g r o u d T r c ) n a f fi as video streaming and VoIP application. Emerging wireless a standards like 802.11e [1] have stronger support for Qual- o g e ity of Service. They differentiate the classes such as Voice, V n a V o g e o e n a h n o e Video, Data, and Background according to the priority. P A d t e r P h n p VoIP provides a low cost alternative to traditional tele- a phone services as they depend only on the broadband lines that users already have, so no additional circuit switching lines are required. VoIP has no physical boundaries and also Figure 1: Vonage equipments setup provides portability. According to [2], VoIP service is ex- pected to grow to over 26M homes in 2008 from 6.5M in 2004. VoIP promises to provide better audio quality primar- 2.1 Setup ily due to higher audio sampling rate when compared to tra- We describe the Vonage equipments setup in Figure 1. ditional telephones [5]. Unlike any other VoIP applications, Vonage service requires Vonage [3] is a market leader in VoIP domain. A keynote phone adapter. It turns broadband connection into a broad- study voted Vonage as the Overall Most Reliable VoIP Ser- band phone line and connects uers to the Vonage network. 1 o n a g e S e r v e r V shake mechanism for establishment connection. By sending an INVITE request, UA1 initiate the connection. UA2 re- sponses back to UA1 by sending 180 Ringing and answering the phone, it also sends 200 OK response back to UA1. Af- ter establishing connection both UAs can trasmitting voice data over the network, at the end of the data exchange any UA can send BYE request to the other for termination. By responding with 200 OK, they can terminate the connection. IN T E R N E T 2.3 RTP and RTCP Different from other peer-to-peer VoIP application, Von- o n a g e V o n a g e V age service maintain central server in-between two users. h o n e P h o n e P Every voice packet from user goes through the intermedi- ate Vonage server and the server relays voice packets to the other user. For the voice data transmission Vonage service uses Real Time Protocol(RTP). The vonage server also sends Figure 2: Every Vonage packet goes through Vonage the Real Time Control Protocol(RTCP) packets to the user server which providesout-of-bandcontorolinfomation for RTP flow between user and server. Vonage server periodically(every 5 seconds) sends control pakcets to participants in a streaming Vonage phone adapter converts voice into data and sends it multimedia session. RTCP gathers statistics on a media con- through the Internet like an email. It also periodically sends nection and information such as bytes sent, packets sent, lost to the Vonage server a register packet which includes user packets, jitter, feedback and round trip delay. information such as IP address and user name to identify the location of the user. 3. IMPACT OF UDP FLOWS ON VONAGE A 1 A 2 U U Vonage service shares the Internet connection with other home network application such as multi media streaming, N V I T E I HTTP and FTP. The quality of the VoIP call traffic is af- fected by the other application due to the limited Internet 1 8 0 R i n g i n g connection bandwidth. For example, people can download 0 0 A C K 2 data files while they use VoIP applications. This might cause the interference between VoIP service and data transmission. O K In this section we identify the impact of other network appli- cations on Vonage service and vice versa. UDP flows doesn’t i a S i o n M e d e s s require ACK for a successful data transmission. We gener- Y E B ate UDP packets for backgroundtraffic while we use Vonage service and present how the background UDP packet impact 0 0 A C K 2 on the quality of Vonage service. We run UDP flows at both ends of the Vonage call. Due to a huge difference (1.5 Mbps and 10 Mbps) in the uplink bandwidth available, the data rate of the UDP flows also grossly different. We report data from the end with the higher bandwidth(10 Mbps). We ob- Figure 3: SIP message flow served significant impact on this end, while the quality on the other end was unaffected, the reason for which is still an open questions. 2.2 Session Initiation Protocol(SIP) Vonage service uses Session Initiation Protocol(SIP) for 3.1 Packet size the connection. SIP is a signaling protocol that is used by Vonage service uses 160 bytes long RTP packets for voice technology for creating session-oriented connection between traffic. We vary the UDP packet size from 100 bytes to 400 two or more endpoints in an IP network. It is an appli- bytes to see how the Vonage router mechanism handles dif- cation layer control protocol that can be used to establish, ferent size of packet and the impact of this mechanisms on modify, and terminate multimedia sesstions. SIP is based Vonage with respect to the packet losses and jitter. From on the peer-to-peer protocoal and the peers are called user- the experiments result Figure 4 we can see that in case of agents(UAs). We describe how the typical SIP messages the UDP packet size is smaller than 150 Bytes, the jitter sig- flow between two UAs in Figure 3. It uses three way hand- nificantly decreases. When background packet size is larger 2 140 than 150 bytes, the average jitter is less than 10 ms, however, UDP size 200Bytes UDP size 700Bytes when packet size is 100 bytes the average jitter is around100 UDP size 1200Bytes 120 ms. When the jitter is high, the quality of Vonage service is bad, sometimes we can’t talk to over the phone clearly. To 100 clarify the packet handling mechanism, we break down the packet size from 150 to 200 bytes, in case of 180 bytes back- 80 ground traffic there is small increases in jitter but doesn’t 60 really affect the quality of voice. We can see that smaller Avg Jitter(ms) packets are handled ahead of the larger packet in router, this 40 causes significant increases in jitter. We need to figure out 20 the point where the jitter is decreases in-between 100 and 0 150 bytes. 0 2 4 6 8 10 12 UDP Data Rate(Mbit/s) 50 Avg Jitter Figure 5: Average Jitter with respect to UDP size 40 packet size of the background traffic is 700-bytes, the jitter 30 is stable around 10 ms for all UDP data rate. We need to run more experiment to figure out this can be happen all the 20 Avg.