a. Draw a network diagram, showing how a telephone in the US would make calls to a telephone on Deception Island. (15 points).

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Pierce Flowers
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1 TSM 350 IP Telephony Fall 2004 E Eichen Exam 1 (Midterm): November 10 Solutions 1 True or False: a Call signaling in a SIP network is routed on a hop-by-hop basis, while call signaling in an H323 network (GK direct model) is be routed directly between VoIP endpoints (5 pts) True or False answer: True Just as it says proxies forward signaling messages hop by hop, where as H323 Gatekeeper Direct, endpoints signal directly to each other after resolving the endpoint address using RAS through gatekeeper(s) b Media (RTP) in a SIP network is routed on a hop-by-hop basis, while media in an H323 network (GK direct model) is be routed directly between VoIP endpoints (5 pts) True or False answer: False Media is typically endpoint-to-endpoint in both H323 and SIP The only exceptions are where there is an intermediate point (such as in a Session Border Controller) where the media is terminated an re-launched 2 Background: Under The Radar Telecom (UTRT Inc) is providing long distance calling, through a calling card server, between the US and The South Shetland Islands Their VoIP Gateway in the US has PSTN connectivity via a channelized T1 to Northeastern's PBX; the US VoIP Gateway also has an Ethernet connection connection to Northeastern s LAN The calling card server, which is the Asterisk PBX in the VoIP lab, is also connected to Northeastern s LAN On Deception Island, UTRT has purchased a fractional T1 from Penguin Net for WAN connectivity to their VoIP gateway in Port Foster The VoIP gateway connects has analog loops to the PSTN switch in Port Foster UTRT is rate limited to ½ of the full clear channel bandwidth on the clear channel T1 from Penguin Net UTRT has decided to use the G723 codec (53 kb/s) because of its low bandwidth, and the fact that most of their customers are penguins who talk slowly As Penguins don t like or web surfing, you can assume that voice RTP packets are the only traffic on the Penguin Net WAN a Draw a network diagram, showing how a telephone in the US would make calls to a telephone on Deception Island (15 points) b How many analog loops should UTRT purchase from the PSTN provider on Deception Island (Hint calculate the bandwidth for a single voice connection, and then divide the total WAN bandwidth by the bandwidth for a single session ) - (20 points) TSM350 Midterm exam p 1

2 Calling Card Server NEU PBX Channel T1 VoIP GW NEU LAN Telephone Public Internet POTS Telephone Port Foster PSTN Switch POTS POTS VoIP GW Figure of UTRT network and service Penguin Net b: answer One half of a T1 is 144Mbits/sec divided by 2, or 720 Kbits/sec The ip packet size will be 53kb/s * 10msec = 53 bits, or 7 bytes, along with a 20 byte IP header, an 8 byte UDP header, 12 bytes for RTP, and 8 bytes for layer 2 headertrailer => 55 bytes The bandwidth required is thus 55 bytes x 8 bits/byte / 10 msec = 44 kbits/sec The number of 44 kb/s channels that can fit inside 720 kb/s is 720/44 = 1636 Thus, the fractional T1 can support 16 phone conversations, and they must buy 16 analog loops Figure of the Layer2 frame for question 2 all numbers are in bytes (octets for the purists) c EEichen, Northeastern CIO (and Vice-President of Greed) has learned of UTRT s service from his contacts at the NSA Rather than shutting down the service, he has asked UTRT to pay for bandwidth, based on the peak bandwidth used during a month Dr Eichen will charge UTRT $1 each month for each kbits/s of peak bandwidth (made payable to a numbered account in Switzerland) Assuming that, at peak, all voice channels on Deception Island are full, how much money will Dr Eichen be collection from UTRT? (Hint this is a little tricky, think about how much bandwidth is required on the NEastern LAN) (20 points) c: answer So, this is a little tricky If the media is terminated by the asterisk server, then there 2x the ethernet media on the LAN If the media is mapped directly to the VoIP GW in Port Foster (more likely) using either a REFER or a RE-INVITE (because the call must first be sent to the calling card server to collect the pin digits), then it s only 16 (per solution to b) simultaneous calls x bandwidth per call For ethernet, we remove the 8 bytes of layer 2, and add 18 bytes for ethernet => 65 byte TSM350 Midterm exam p 2

3 packets The peak bandwidth is thus 16*65 bytes * 8 bits/byte/10 msec = 832 kbits/s I am thus going to make $832/month Not bad 3 Please draw a ladder diagram based on the attached diagram and a snippet of SIP debugs What s going on here? (35 points) Router Ethernet Switch IP Phone ext IP Phone ext 3002 Network Diagram for problem Asterisk Server So what s going on here is that 101 (ext 3004) is calling 105 (ext 3002), but the media and signaling are being forced through the PBX The phone at 3002 is busy (in this case, I set it to Do Not Disturb), and the voice mail is being left on the server and a NOTIFY message is sent to the phone (3002) to turn on the message waiting indicator light Here s the ladder diagram: ext Asterisk ext 3002 INVITE TRYING INVITE BUSY OK ACK ACK RTP (although, you can't see this from the SIP debug) NOTIFY OK BYE OK TSM350 Midterm exam p 3

4 U :5060 -> :5060 INVITE SIP/20 Via: SIP/20/UDP :5060 From: To: Call-ID: CSeq: 2 INVITE Contact: <sip:3004@ :5060;user=phone;transport=udp> User-Agent: Cisco ATA 186 v310 atasip (040211A) Proxy-Authorization: Digest username="3004",realm="asterisk",nonce="1fdb7e7b",uri="sip:3002@ ",respons e="a1bb3a7501b a93bffacca0568" Expires: 300 Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER Content-Length: 247 Content-Type: application/sdp v=0 o= IN IP s=ata186 Call c=in IP t=0 0 m=audio RTP/AVP a=rtpmap:0 PCMU/8000/1 a=rtpmap:4 G723/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp: U :5060 -> :5060 SIP/ Trying Via: SIP/20/UDP :5060 From: <sip:3004@ ;user=phone>;tag= To: <sip:3002@ ;user=phone>;tag=as6ce511fd Call-ID: @ CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:3002@ > Content-Length: 0 U :5060 -> :5060 INVITE sip:3002@ SIP/20 Via: SIP/20/UDP :5060;branch=z9hG4bK06dae12e From: "3004" <sip:3004@ >;tag=as39b4da84 To: <sip:3002@ > Contact: <sip:3004@ > Call-ID: 637e ad07d4e39a7e6ef404@ CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Sun, 14 Nov :19:49 GMT Alert-info: Bellcore-dr1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 238 v=0 o=root IN IP s=session c=in IP t=0 0 m=audio RTP/AVP TSM350 Midterm exam p 4

5 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp: U : > :5060 SIP/ Busy here Via: SIP/20/UDP :5060;branch=z9hG4bK06dae12e From: "3004" <sip:3004@ >;tag=as39b4da84 To: <sip:3002@ >;tag=000628f0f9f2000d7a71ad55-38f1a626 Call-ID: 637e ad07d4e39a7e6ef404@ Date: Sun, 14 Nov :19:49 GMT CSeq: 102 INVITE Server: CSCO/4 Contact: <sip:3005@ :5060> Content-Length: 0 U :5060 -> :5060 ACK sip:3005@ :5060 SIP/20 Via: SIP/20/UDP :5060;branch=z9hG4bK06dae12e From: "3004" <sip:3004@ >;tag=as39b4da84 To: <sip:3002@ >;tag=000628f0f9f2000d7a71ad55-38f1a626 Contact: <sip:3004@ > Call-ID: 637e ad07d4e39a7e6ef404@ CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 U :5060 -> :5060 SIP/ OK Via: SIP/20/UDP :5060 From: <sip:3004@ ;user=phone>;tag= To: <sip:3002@ ;user=phone>;tag=as6ce511fd Call-ID: @ CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:3002@ > Content-Type: application/sdp Content-Length: 237 v=0 o=root IN IP s=session c=in IP t=0 0 m=audio 5812 RTP/AVP a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp: U :5060 -> :5060 ACK sip:3002@ SIP/20 Via: SIP/20/UDP :5060 TSM350 Midterm exam p 5

6 From: To: Call-ID: CSeq: 2 ACK User-Agent: Cisco ATA 186 v310 atasip (040211A) Proxy-Authorization: Digest username="3004",realm="asterisk",nonce="1fdb7e7b",uri="sip:3002@ ",respons e="a1bb3a7501b a93bffacca0568" Content-Length: 0 U :5060 -> :5060 NOTIFY sip:3002@ SIP/20 Via: SIP/20/UDP :5060;branch=z9hG4bK361f812f From: "asterisk" <sip:asterisk@ >;tag=as39e043e0 To: <sip:3002@ > Contact: <sip:asterisk@ > Call-ID: 3a86cecd339da70710f761f11e0515d4@ CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 37 Messages-Waiting: yes Voic 2/0 U : > :5060 SIP/ OK Via: SIP/20/UDP :5060;branch=z9hG4bK361f812f From: "asterisk" <sip:asterisk@ >;tag=as39e043e0 To: <sip:3002@ > Call-ID: 3a86cecd339da70710f761f11e0515d4@ Date: Sun, 14 Nov :20:04 GMT CSeq: 102 NOTIFY Content-Length: 0 U :5060 -> :5060 BYE sip:3004@ :5060 SIP/20 Via: SIP/20/UDP :5060;branch=z9hG4bK216fee2d From: <sip:3002@ ;user=phone>;tag=as6ce511fd To: <sip:3004@ ;user=phone>;tag= Contact: <sip:3002@ > Call-ID: @ CSeq: 102 BYE User-Agent: Asterisk PBX Content-Length: 0 U :5060 -> :5060 SIP/ OK Via: SIP/20/UDP :5060;branch=z9hG4bK216fee2d From: <sip:3002@ ;user=phone>;tag=as6ce511fd To: <sip:3004@ ;user=phone>;tag= Call-ID: @ CSeq: 102 BYE Server: Cisco ATA 186 v310 atasip (040211A) Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER Content-Length: 0 TSM350 Midterm exam p 6

TSM350G Midterm Exam MY NAME IS March 12, 2007

TSM350G Midterm Exam MY NAME IS March 12, 2007 PLEAE PUT ALL YOUR ANSWERS in a BLUE BOOK with YOUR NAME ON IT IF you are using more than one blue book, please put your name on ALL blue books 1 Attached