Sip and sdp relationship

Understanding Session Description Protocol (SDP) | SIP Adventures

sip and sdp relationship

SIP Session Description Protocol - Learn Session Initiation Protocol in simple SDP is generally contained in the body part of Session Initiation Protocol popularly called SIP. The c= field contains information about the media connection. SDP, also known as Session Description Protocol is the protocol used with SIP and to use different transport protocols as appropriate, such as SIP and HTTP. c= (connection information – not required if included in media description)*. SIP does what it does best and leaves media to SDP. So, what c=* (connection information — optional if included at session level). b=* (zero.

When B accepts the call his user agent sends a message with a response code of Any 2xx response means that the message was successfully received, understood, and accepted.

The final part of the three-way handshake occurs when A sends an acknowledgement to B. By sending an ACK the caller confirms that it has received the response from the callee. After the setup procedure is completed the conversation can begin now using RTP.

sip and sdp relationship

SIP protocol is used to initiate a session between two endpoints: In Asterisk it is possible to debug all the SIP messages with the following commands from console. We have see that the SIP protocol can be, and usually is, routed through one or more SIP proxy servers before reaching its destination: Each email server adds a Received header to the message, to track the route the message has taken.

The Via field indicates the path taken by the request so far.

RE: [Sipping] Relationship between SIP session and SDP session

This prevents request looping and ensures replies take the same path as the requests, which assists in firewall traversal and other unusual routing situations. The audio component obviously signifies that this is an audio call, specifies the port where want to receive the RTP stream, and the IP address is specified in 6: The numbers at the end of this header represent the different codecs that this client supports: But some companies use values of 10, 30, or 40 ms.

Ptime is an important consideration in your VoIP deployment. The amount of audio captured in each packet affects the impact on your audio quality if an individual packet is lost, and on how many packets per second your servers must handle to pass the same amount of audio. Figure illustrates how the ptime you choose for your transmission impacts your network.

The transmission requires 20 packets at a sampling rate of 10 ms each to successfully transmit one-tenth of a second of audio ms. A simple increase in sampling size to 20 ms cuts the quantity of packets required to send the same ms of audio in half.

  • SIP - Session Description Protocol
  • Session Description Protocol
  • Understanding Session Description Protocol (SDP)

The increase in volume can easily overload your routers, creating an increase in lost packets. This kind of systemic congestion can clog your network, cause poor call quality, and lead to completion errors on call attempts.

sip and sdp relationship

You need to pay attention to packet volume because routers and LAN equipment are rated based on the quantity of packets per second they can handle. So, if small packets cause congestion problems, you can choose the largest packet size available. You can effectively cut the quantity of packets in half if you go from a ms packet to a ms packet.

Reducing half of the packets running through your LAN or WAN should make it run smoothly because it now takes only five packets to transmit the same ms of audio. But when you use a large packet, you have to accept a trade-off. The larger ms packets are carrying quite a bit of audio for your call. If a single ms packet is lost, or arrives out of sequence at the receiving router at the far end of the call and has to be thrown out, the people engaged in the call notice.

The technical term for when a packet is dropped is clipping. The larger the packet, the more damage packet loss does to the audio of the call, even if only a single packet is lost.

If the same quantity of packets were lost, but the sampling rate was 20 ms, the impact on the call quality might be negligible.

Each message may contain multiple timing and media descriptions.

Session Description Protocol - Wikipedia

Names are only unique within the associated syntactic construct, i. This session is originated by the user 'jdoe', at IPv4 address Its name is "SDP Seminar" and extended session information "A Seminar on the session description protocol" is included along with a link for additional information and an email address to contact the responsible party, Jane Doe.

This session is specified to last for two hours using NTP timestamps, with a connection address which indicates the address clients must connect to or — when a multicast address is provided, as it is here — subscribe to specified as IPv4 Recipients of this session description are instructed to only receive media. RTCP ports for the audio and video streams of andrespectively, are implied.

Attributes[ edit ] SDP uses attributes to extend the core protocol. Attributes can appear within the Session or Media sections and are scoped accordingly as session-level or media-level.