A phone call

  Want to know what a phone call over the internet can look like? Consider that the following is simply the initial INVITE message which is basically the call offer. This doesn't show the extra messaging that goes on for codec negotiation, any call progress or ringing indication, when the call is actually answered and connected, when the call hangs up, when the call can be transferred etc. You get the idea. Anyway on to the message itself (this is the SIP protocol FYI).


INVITE sip:+15105551005@vxst.com;user=phone SIP/2.0
Via: SIP/2.0/TLS 10.10.225.170:5061;branch=z9hG4bK14e473a9a4d8;tdmchannel="b:1 t:1 g:1 c:23"
Max-Forwards: 70
To:
From: "System Test" ;tag=36e30006
Call-ID: 6-6d5420091118174918421@10.10.225.170
CSeq: 178131676 INVITE
Contact: ;isGateway;transport=TLS
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, REFER, NOTIFY, REGISTER, PRACK, SUBSCRIBE, UPDATE
Allow-Events: message-summary, talk, vq-rtcpxr
Supported: 100rel, replaces
P-Asserted-Identity: "System Test"
Remote-Party-ID: "System Test" ;party=calling;privacy=off;npi=5
Cisco-Guid: 2524999362-786452308-2434106206-2147673518
User-Agent: My user agent
Content-Length: 2388
Content-Type: multipart/alternative;boundary="---Alt-1ca6877-73a9a4d8-5"

-----Alt-1ca6877-73a9a4d8-5
Content-Type: application/sdp
Content-Disposition: session; handling=optional; ms-proxy-2007fallback

v=0
o=SHOUT 0 0 IN IP4 10.10.225.170
s=SIP call via Me
c=IN IP4 10.10.225.170
t=0 0
m=audio 16384 RTP/AVP 97 114 115 0 13 101
a=candidate:Ddupc9byqXO5vqlzOY2pc3S1qXOC8KlzlYOpczKlqXM= 1 Koupc1uOqXNX5KlzObKpcw== UDP 0.900 10.10.225.170 16384
a=candidate:Ddupc9byqXO5vqlzOY2pc3S1qXOC8KlzlYOpczKlqXM= 2 Koupc1uOqXNX5KlzObKpcw== UDP 0.900 10.10.225.170 16385
a=candidate:spypc23iqXPaoKlzJb2pc8q/qXNsrKlzudupc9fCqXM= 1 m/Wpc3XEqXNZ46lzpbqpcw== UDP 0.800 10.51.100.8 50238
a=candidate:spypc23iqXPaoKlzJb2pc8q/qXNsrKlzudupc9fCqXM= 2 m/Wpc3XEqXNZ46lzpbqpcw== UDP 0.800 10.51.100.8 55746
a=rtcp:16385
a=rtpmap:97 red/8000
a=fmtp:97 114/114
a=rtpmap:114 x-msrta/8000
a=fmtp:114 bitrate=8800
a=rtpmap:115 x-msrta/8000
a=fmtp:115 bitrate=11800
a=rtpmap:0 PCMU/8000
a=rtpmap:13 CN/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:Mo+HeCAc/zxNQFHcCYFUfxdbNslYv+wdQmqO602R|2^31|1:1 WSH=64
a=cryptoscale:2 client AES_CM_128_HMAC_SHA1_80 inline:2s8CdsuYukIPJziMIJHDrActSwn4g73KBQS7m41d|2^31|2:1 WSH=64
a=ptime:20
a=tcap:1 RTP/SAVP
a=pcfg:1 t=1

-----Alt-1ca6877-73a9a4d8-5
Content-Type: application/sdp
Content-Disposition: session; handling=optional

v=0
o=SHOUT 0 0 IN IP4 10.10.225.170
s=SIP call via Me
c=IN IP4 10.10.225.170
t=0 0
m=audio 16384 RTP/AVP 97 114 115 0 13 101
a=ice-ufrag:w5Cp
a=ice-pwd:HZOpc0G+qXPa96lzle+pcw
a=candidate:1 1 UDP 2130706431 10.10.225.170 16384 typ host
a=candidate:1 2 UDP 2130706430 10.10.225.170 16385 typ host
a=candidate:2 1 UDP 16777215 10.51.100.8 50238 typ relay raddr 10.51.100.8 rport 50238
a=candidate:2 2 UDP 16777215 10.51.100.8 55746 typ relay raddr 10.51.100.8 rport 55746
a=rtcp:16385
a=rtpmap:97 red/8000
a=fmtp:97 114/114
a=rtpmap:114 x-msrta/8000
a=fmtp:114 bitrate=8800
a=rtpmap:115 x-msrta/8000
a=fmtp:115 bitrate=11800
a=rtpmap:0 PCMU/8000
a=rtpmap:13 CN/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:Mo+HeCAc/zxNQFHcCYFUfxdbNslYv+wdQmqO602R|2^31|1:1 WSH=64
a=cryptoscale:2 client AES_CM_128_HMAC_SHA1_80 inline:2s8CdsuYukIPJziMIJHDrActSwn4g73KBQS7m41d|2^31|2:1 WSH=64
a=ptime:20
a=tcap:1 RTP/SAVP
a=pcfg:1 t=1

-----Alt-1ca6877-73a9a4d8-5--


Just thought I would show this since there is a lot more going on behind the scenes when you make that simple phone call that goes over the internet. BTW, most cell phones actually make calls that effectively go over the internet now from a protocol point of view, anything on 3G and above basically sets up a stream of data, no more fixed bandwidth lines.

  Phones have come a long way in the last 100 years or so.