Following is the complete process for making a call:
INVITE MESSAGE:
INVITE sip:30001dmojs@192.168.1.103:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 64.34.231.248:5060;branch=z9hG4bK2ef7805b;rport
From: "Reception"
To:
Contact:
Call-ID: 400da942638ea5c96724784846686f7d@64.34.231.248
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 18 Aug 2009 14:49:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 242
v=0
o=root 19191 19191 IN IP4 64.34.231.248
s=session
c=IN IP4 64.34.231.248
t=0 0
m=audio 13354 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
| INVITE sip:30001dmojs@192.168.1.103:5060;transport=udp SIP/2.0 | |
| Via: SIP/2.0/UDP 64.34.231.248:5060;branch=z9hG4bK2ef7805b;rport | |
| From: "Reception" | |
| To: | |
| Contact: | |
| Call-ID: 400da942638ea5c96724784846686f7d@64.34.231.248 | |
| CSeq: 102 INVITE | |
| User-Agent: Asterisk PBX | |
| Max-Forwards: 70 | |
| Date: Tue, 18 Aug 2009 14:49:15 GMT | |
| Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY | |
| Supported: replaces | |
| Content-Type: application/sdp | |
| Content-Length: 242 | |
| Session Description Protocol · (v): SDP version. The preceding section’s example code uses version 0. · (o): Owner/creator and session ID. · (s): Session name. The example session isn’t named anything fancy, just branded with a default value of session. · (c): Connection information. This identifies the version of Internet Protocol software used to generate the packet, along with the IP address from which the SDP was transmitted. · (t): The time the session is active. | |
| v=0 | |
| o=root 19191 19191 IN IP4 64.34.231.248 | Owner name or the sip server name is root |
| | Network time protocol (NTP) timestamp determines the time. A standard packet of voice generally represents only 20 milliseconds of sound. If NTP had any more variance than it does, transmissions could easily end up out of sequence. |
| | IN is the internet |
| | IP4 is the version used |
| | 64.34.231.248 is the source address |
| s=session | Session name |
| c=IN IP4 64.34.231.248 | Can be different from above if separate server is used for media server |
| t=0 0 | Start and stop time of a conference session. |
| Media level description | |
| m=audio 13354 RTP/AVP 0 101 | Audio transmission is used at the port 13354 to transmit the media of the call. Unique to each call. Real Time Protocol and Audio/Video Profile to be used during the call 0[HC1] refer to RTP payload types which correspond to the codecs. 0 is ulaw codec 101 describes this is a telephone event |
| a further defines the RTP payload types | |
| a=rtpmap:0 PCMU/8000 | This tells to use the uLaw codec at the sampling rate of 8000 Hz so that both ends sample at same rate |
| a=rtpmap:101 telephone-event/8000 | |
| a=fmtp:101 0-16 | Dtmf tones enabled (0,1,2,….., A,B,…, #) |
| a=silenceSupp:off - - - - | If silence suppression is on, no data is transmitted during the call if no one is talking |
| a=ptime:20 | 20 of voicems is stored in each packet |
| a=sendrecv | |
The SDP information in the 200 OK response that’s sent against the example INVITE looks like this:
Media Description, name and address (m): audio 52150 RTP/
AVP 18 101
Media Attribute (a): ptime: 20
Media Attribute (a): rtpmap: 101 telephone-event/8000
Media Attribute (a): fmtp: 101 0-15
Media Attribute (a): sendrecv
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