TSM350G Midterm Exam MY NAME IS March 12, 2007

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1 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 are the sip messages for a call to my home asterisk server a Draw the ladder diagram (35 points) b What is happening here, what is the end user doing (5 points) 2 Provide three functions that a Session Border Controller provides (one sentence each) (10 points each) a b c 3 You have been asked to provide voice over IP to a small branch office that has a single T1 for IP connectivity The customer wants to use G722 as the audio codec The customer also needs to reserve about ½ of the bandwidth on the T1 for data, and thus is willing to devote about ½ the bandwidth on the T1 for voice {By the way, in reality, it s never a good idea to plan on using 100% of the bandwidth of an IP link, even one that is only providing voip A good rule of thumb is to use a max of 80% of the bandwidth on a link; otherwise, the burstiness of the of the IP data will cause some data to get thrown away, which in turn will create re-transmissions of the TCP stuff, which in turn will add more data to the connection, which in turn will cause congestion and a hosed connection However, for the purposes of this exam, assume that it s ok to go to 100%!) G722 will provide 48,56, or 64 Kbits/s You have chosen 56 kbits/s, along with packing 40 msec of data into a single RTP frame How many voice calls will you be able to support on the ½ T1 (30 points) You may assume Cisco HDLC for the layer 2 protocol, which requires 5 bytes of header data and 3 bytes of trailer data (pretty efficient!) You may also assume that the only significant bandwidth used in a voip call is RTP (eg, the SIP signaling, DNS quires, RTCP control, etc use negligible bandwidth) TSM G350 midterm March p 1

2 Script started on Sun 11 Mar :29:21 PM EDT elliot] ngrep -W byline host and port 5060 or host and port 5060 interface: eth0 ( / ) filter: (ip) and ( host and port 5060 or host and port 5060 ) U :5060 -> :2051 INVITE sip:3002@ :2051;line=g7vcmvc1 SIP/20 Via: SIP/20/UDP :5060;branch=z9hG4bK446542df From: "Unknown" <sip:unknown@ >;tag=as14c2524f To: <sip:3002@ :2051;line=g7vcmvc1> Contact: <sip:unknown@ > Call-ID: 49a2e8fd4ea5f d1e @ CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sun, 11 Mar :29:53 GMT Alert-Info: Bellcore-dr1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 240 v=0 o=root IN IP s=session c=in IP t=0 0 m=audio RTP/AVP a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp: a=silencesupp:off U :2051 -> :5060 SIP/ Ringing Via: SIP/20/UDP :5060;branch=z9hG4bK446542df From: "Unknown" <sip:unknown@ >;tag=as14c2524f To: <sip:3002@ :2051;line=g7vcmvc1>;tag=jrzru2qbuc Call-ID: 49a2e8fd4ea5f d1e @ CSeq: 102 INVITE Contact: <sip:3002@ :2051;line=g7vcmvc1>;flow-id=1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0 U :2051 -> :5060 SIP/ Ok Via: SIP/20/UDP :5060;branch=z9hG4bK446542df From: "Unknown" <sip:unknown@ >;tag=as14c2524f To: <sip:3002@ :2051;line=g7vcmvc1>;tag=jrzru2qbuc Call-ID: 49a2e8fd4ea5f d1e @ CSeq: 102 INVITE Contact: <sip:3002@ :2051;line=g7vcmvc1>;flow-id=1 User-Agent: snom360/55 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Content-Type: application/sdp Content-Length: 204 v=0 o=root IN IP s=call TSM G350 midterm March p 2

3 c=in IP t=0 0 m=audio RTP/AVP a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp: a=sendrecv U :5060 -> :2051 ACK sip:3002@ :2051;line=g7vcmvc1 SIP/20 Via: SIP/20/UDP :5060;branch=z9hG4bK4db7c34a From: "Unknown" <sip:unknown@ >;tag=as14c2524f To: <sip:3002@ :2051;line=g7vcmvc1>;tag=jrzru2qbuc Contact: <sip:unknown@ > Call-ID: 49a2e8fd4ea5f d1e @ CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 U :2051 -> :5060 REFER sip:unknown@ SIP/20 Via: SIP/20/UDP :2051;branch=z9hG4bK-jvedgcsprzy8;rport From: <sip:3002@ :2051;line=g7vcmvc1>;tag=jrzru2qbuc To: "Unknown" <sip:unknown@ >;tag=as14c2524f Call-ID: 49a2e8fd4ea5f d1e @ CSeq: 2 REFER Max-Forwards: 70 Contact: <sip:3002@ :2051;line=g7vcmvc1>;flow-id=1 Refer-To: sip:3004@ ;user=phone Referred-By: sip:3002@ User-Agent: snom360/55 Content-Length: 0 U :5060 -> :2051 SIP/ Accepted Via: SIP/20/UDP :2051;branch=z9hG4bKjvedgcsprzy8;rport;received= From: <sip:3002@ :2051;line=g7vcmvc1>;tag=jrzru2qbuc To: "Unknown" <sip:unknown@ >;tag=as14c2524f Call-ID: 49a2e8fd4ea5f d1e @ CSeq: 2 REFER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:unknown@ > Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing U :5060 -> :2051 BYE sip:3002@ :2051;line=g7vcmvc1 SIP/20 Via: SIP/20/UDP :5060;branch=z9hG4bK464a684b From: "Unknown" <sip:unknown@ >;tag=as14c2524f To: <sip:3002@ :2051;line=g7vcmvc1>;tag=jrzru2qbuc Contact: <sip:unknown@ > Call-ID: 49a2e8fd4ea5f d1e @ CSeq: 104 BYE User-Agent: Asterisk PBX Max-Forwards: 70 X-Asterisk-HangupCause: Normal Clearing Content-Length: 0 TSM G350 midterm March p 3

4 U :5060 -> :5060 INVITE SIP/20 Via: SIP/20/UDP :5060;branch=z9hG4bK61da7760 From: "Unknown" To: Contact: Call-ID: CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sun, 11 Mar :30:18 GMT Alert-Info: Bellcore-dr1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 240 v=0 o=root IN IP s=session c=in IP t=0 0 m=audio RTP/AVP a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp: a=silencesupp:off U :5060 -> :5060 SIP/ Trying Via: SIP/20/UDP :5060;branch=z9hG4bK61da7760 From: "Unknown" <sip:unknown@ >;tag=as73b47d45 To: <sip:3004@ :5060;user=phone;transport=udp>;tag= Call-ID: 0be303d3707ca68047a5e4046c5c1a2c@ CSeq: 102 INVITE Server: Cisco ATA 186 v310 atasip (040211A) Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER Content-Length: 0 U :5060 -> :5060 SIP/ Ringing Via: SIP/20/UDP :5060;branch=z9hG4bK61da7760 From: "Unknown" <sip:unknown@ >;tag=as73b47d45 To: <sip:3004@ :5060;user=phone;transport=udp>;tag= Call-ID: 0be303d3707ca68047a5e4046c5c1a2c@ CSeq: 102 INVITE Server: Cisco ATA 186 v310 atasip (040211A) Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER Content-Length: 0 U :2051 -> :5060 SIP/ OK Via: SIP/20/UDP :5060;branch=z9hG4bK464a684b From: "Unknown" <sip:unknown@ >;tag=as14c2524f To: <sip:3002@ :2051;line=g7vcmvc1>;tag=jrzru2qbuc Call-ID: 49a2e8fd4ea5f d1e @ CSeq: 104 BYE Contact: <sip:3002@ :2051;line=g7vcmvc1>;flow-id=1 User-Agent: snom360/55 Content-Length: 0 TSM G350 midterm March p 4

5 U :5060 -> :5060 SIP/ OK Via: SIP/20/UDP :5060;branch=z9hG4bK61da7760 From: "Unknown" To: Call-ID: CSeq: 102 INVITE Contact: Server: Cisco ATA 186 v310 atasip (040211A) Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER Content-Length: 195 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:101 telephone-event/8000 a=fmtp: U :5060 -> :5060 ACK sip:3004@ :5060;user=phone;transport=udp SIP/20 Via: SIP/20/UDP :5060;branch=z9hG4bK1363bed5 From: "Unknown" <sip:unknown@ >;tag=as73b47d45 To: <sip:3004@ :5060;user=phone;transport=udp>;tag= Contact: <sip:unknown@ > Call-ID: 0be303d3707ca68047a5e4046c5c1a2c@ CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 U :5060 -> :5060 BYE sip:unknown@ SIP/20 Via: SIP/20/UDP :5060 From: <sip:3004@ ;user=phone;transport=udp>;tag= To: "Unknown" <sip:unknown@ >;tag=as73b47d45 Call-ID: 0be303d3707ca68047a5e4046c5c1a2c@ CSeq: 1 BYE User-Agent: Cisco ATA 186 v310 atasip (040211A) Content-Length: 0 U :5060 -> :5060 SIP/ OK Via: SIP/20/UDP :5060;received= From: <sip:3004@ ;user=phone;transport=udp>;tag= To: "Unknown" <sip:unknown@ >;tag=as73b47d45 Call-ID: 0be303d3707ca68047a5e4046c5c1a2c@ CSeq: 1 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:unknown@ > Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing TSM G350 midterm March p 5

6 Problem 1a: Ladder diagram INVITE RINGING OK ext Ext 3004 ACK REFER ACCEPT BYE INVITE TRYING RINGING OKOK OK OK ACK BYE OK OK Problem 1b The called party at extension 3002 TRANSFERS (or diverts) the call to extension 3004 Lots of people thought this was about 3 rd party call control since it uses a REFER method In this case it s not 3 rd party cc (need to review this) 2: Lots of answers Here I was looking for some basic understanding, AND NOT SOMETHING THAT SOMEONE HAD COPIED DIRECTLY OUT OF A Vendor s material There were many folks (>4) that had identical statements about gracefully rejecting calls, and other such stuff Also, lots of folks simply inundated me with words about 14 functions in the same line, or put down additional functionality in lines d,e,f, sorry, this doesn t work, it s not a lottery where you get to try this one, and if not this one then that one, or how about or maybe you will believe a NAT traversal, end-point registration, infrastructure hiding, 911 support, CALEA implementations, RTP transcoding, signaling transcoding (SIP to H323, etc), signaling fix up, SIP firewall, routing, blind refile, etc TSM G350 midterm March p 6

7 3: The number of bits in a 40msec slice of 56kbits/sec = 40E-3sec X 56E3 bits/sec = 2240 bits = 2240/8 bits/byte = 280 bytes If RTP adds 12 bytes, UDP adds 8 bytes, IP adds 20 bytes, and layer 2 adds 8 bytes, than the overhead is 48 bytes, and the entire packet is = 328 bytes The bandwidth is thus 328 bytes every 40 msec of = 328bytes X 8 bits/40e3 sec = 656 kbits/sec Since a T1 is 154 MBits/sec full duplex, ½ a T1 is 770 kbits/sec The number of simultaneous voice conversations (channels) that this ½ T1 can support is thus 11 Grades: TSM G350 midterm March p 7

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