rvoip_sip_core::builder::headers::content

Trait ContentBuilderExt

pub trait ContentBuilderExt: ContentTypeBuilderExt {
    // Required method
    fn sdp_body(self, sdp: &SdpSession) -> Self;
}

Expand description

Content Body Builder for SIP Messages

This module provides builder methods for setting message body content in SIP messages, working in conjunction with Content-Type, Content-Length, and other content-related headers.

§SIP Message Content Overview

SIP messages can contain various types of content in their bodies, which is described by the Content-Type header as defined in RFC 3261 Section 20.15. The Content-Length header RFC 3261 Section 20.14 specifies the size of the body.

§Purpose of Message Content

Message bodies in SIP serve several critical functions:

Media negotiation: SDP content in INVITE requests describes media capabilities
Instant messaging: Text or MIME content in MESSAGE requests for IM applications
Event notification: XML or JSON content in NOTIFY requests for events and presence
File transfer: Binary content for delivering files or images
Application data: Custom application data exchange between SIP endpoints

§Common Content Types in SIP

application/sdp: Session Description Protocol for media negotiation
text/plain: Simple text messages
application/xml: XML-based content (PIDF, XCAP, etc.)
application/json: JSON-formatted data
message/sipfrag: SIP message fragments (used in NOTIFY for REFER status)
multipart/mixed: Multiple content parts with different types
image/jpeg, image/png: Image content for avatar updates or visual voicemail

§Relationship with other headers

Content-Type: Specifies the MIME type of the message body
Content-Length: Indicates the size of the body in octets
Content-Encoding: Specifies any compression or encoding applied to the body
Content-Language: Indicates the natural language of the body content
Content-Disposition: Indicates how the body should be interpreted
MIME-Version: Required when using multipart content types

§Examples

§INVITE with SDP for Audio Call

use rvoip_sip_core::builder::SimpleRequestBuilder;
use rvoip_sip_core::builder::headers::ContentBuilderExt;
use rvoip_sip_core::types::Method;
use rvoip_sip_core::sdp::SdpBuilder;

// Create an SDP session for an audio call
let sdp = SdpBuilder::new("Audio Call")
    .origin("alice", "2890844526", "2890844526", "IN", "IP4", "alice.example.org")
    .connection("IN", "IP4", "alice.example.org")
    .time("0", "0")
    .media("audio", 49170, "RTP/AVP")
    .formats(&["0", "8", "96"])
    .attribute("rtpmap", Some("0 PCMU/8000"))
    .attribute("rtpmap", Some("8 PCMA/8000"))
    .attribute("rtpmap", Some("96 telephone-event/8000"))
    .attribute("ptime", Some("20"))
    .done()
    .build()
    .unwrap();

// Create an INVITE request for an audio call
let request = SimpleRequestBuilder::new(Method::Invite, "sip:bob@example.com").unwrap()
    .from("Alice", "sip:alice@example.com", Some("a73kszlfl"))
    .to("Bob", "sip:bob@example.com", None)
    .contact("<sip:alice@192.0.2.1:5060>", None)
    .sdp_body(&sdp)  // Sets Content-Type to application/sdp automatically
    .build();

§Audio and Video Call with SRTP

use rvoip_sip_core::builder::SimpleRequestBuilder;
use rvoip_sip_core::builder::headers::ContentBuilderExt;
use rvoip_sip_core::types::Method;
use rvoip_sip_core::sdp::SdpBuilder;

// Create an SDP session for a secure audio/video call
let sdp = SdpBuilder::new("Secure Audio/Video Call")
    .origin("alice", "2890844526", "2890844526", "IN", "IP4", "alice.example.org")
    .connection("IN", "IP4", "alice.example.org")
    .time("0", "0")
    // Add audio media with SRTP
    .media("audio", 49170, "RTP/SAVP")
    .formats(&["0", "8", "96"])
    .attribute("rtpmap", Some("0 PCMU/8000"))
    .attribute("rtpmap", Some("8 PCMA/8000"))
    .attribute("rtpmap", Some("96 telephone-event/8000"))
    .attribute("crypto", Some("1 AES_CM_128_HMAC_SHA1_80 inline:PS1uQCVeeCFCanVmcjkpPywjNWhcYD0mXXtxaVBR|2^20|1:32"))
    .done()
    // Add video media with SRTP
    .media("video", 51372, "RTP/SAVP")
    .formats(&["97", "98"])
    .attribute("rtpmap", Some("97 H264/90000"))
    .attribute("rtpmap", Some("98 VP8/90000"))
    .attribute("crypto", Some("1 AES_CM_128_HMAC_SHA1_80 inline:d0RmdmcmVCspeEc3QGZiNWpVLFJhQX1cfHAwJSoj|2^20|1:32"))
    .done()
    .build()
    .unwrap();

// Create an INVITE request for a secure audio/video call
let request = SimpleRequestBuilder::new(Method::Invite, "sip:bob@example.com").unwrap()
    .from("Alice", "sip:alice@example.com", Some("secure-call"))
    .to("Bob", "sip:bob@example.com", None)
    .contact("<sip:alice@203.0.113.5:5061;transport=tls>", None)
    .sdp_body(&sdp)  // Sets Content-Type to application/sdp automatically
    .build();

§WebRTC Media Negotiation

use rvoip_sip_core::builder::SimpleRequestBuilder;
use rvoip_sip_core::builder::headers::ContentBuilderExt;
use rvoip_sip_core::types::Method;
use rvoip_sip_core::sdp::SdpBuilder;

// Create an SDP session for WebRTC (with ICE and DTLS)
let sdp = SdpBuilder::new("WebRTC Call")
    .origin("-", "1545997027", "1", "IN", "IP4", "198.51.100.1")
    .time("0", "0")
    .attribute("group", Some("BUNDLE 0 1"))
    .attribute("ice-options", Some("trickle"))
    .attribute("msid-semantic", Some("WMS myWebRTCStream"))
    // Add audio media
    .media("audio", 49203, "UDP/TLS/RTP/SAVPF")
    .formats(&["111", "103", "104", "9", "0", "8"])
    .connection("IN", "IP4", "198.51.100.1")
    .attribute("rtcp", Some("60065 IN IP4 198.51.100.1"))
    .attribute("candidate", Some("1 1 UDP 2113937151 192.168.1.4 49203 typ host"))
    .attribute("candidate", Some("2 1 UDP 1845501695 198.51.100.1 49203 typ srflx raddr 192.168.1.4 rport 49203"))
    .attribute("ice-ufrag", Some("F7gI"))
    .attribute("ice-pwd", Some("x9cml/YzichV2+XlhiMu8rbz"))
    .attribute("fingerprint", Some("sha-256 D2:B9:31:8F:DF:24:D8:0E:ED:D2:EF:25:9E:AF:6F:B8:05:A3:75:A1:A4:1C:6B:1E:55:02:A4:F9:6B:CA:F7:E6"))
    .attribute("setup", Some("actpass"))
    .attribute("mid", Some("0"))
    .attribute("rtpmap", Some("111 opus/48000/2"))
    .attribute("rtcp-fb", Some("111 transport-cc"))
    .attribute("fmtp", Some("111 minptime=10;useinbandfec=1"))
    .attribute("rtpmap", Some("103 ISAC/16000"))
    .attribute("rtpmap", Some("104 ISAC/32000"))
    .attribute("rtpmap", Some("9 G722/8000"))
    .attribute("rtpmap", Some("0 PCMU/8000"))
    .attribute("rtpmap", Some("8 PCMA/8000"))
    .attribute("extmap", Some("1 urn:ietf:params:rtp-hdrext:ssrc-audio-level"))
    .attribute("ssrc", Some("2655508255 cname:ConferenceTool"))
    .attribute("ssrc", Some("2655508255 msid:myWebRTCStream audioTrack"))
    .attribute("ssrc", Some("2655508255 mslabel:myWebRTCStream"))
    .attribute("ssrc", Some("2655508255 label:audioTrack"))
    .done()
    // Add video media
    .media("video", 49203, "UDP/TLS/RTP/SAVPF")
    .formats(&["96", "97", "98", "99", "100", "101", "102"])
    .connection("IN", "IP4", "198.51.100.1")
    .attribute("rtcp", Some("60065 IN IP4 198.51.100.1"))
    .attribute("candidate", Some("1 1 UDP 2113937151 192.168.1.4 49203 typ host"))
    .attribute("candidate", Some("2 1 UDP 1845501695 198.51.100.1 49203 typ srflx raddr 192.168.1.4 rport 49203"))
    .attribute("ice-ufrag", Some("F7gI"))
    .attribute("ice-pwd", Some("x9cml/YzichV2+XlhiMu8rbz"))
    .attribute("fingerprint", Some("sha-256 D2:B9:31:8F:DF:24:D8:0E:ED:D2:EF:25:9E:AF:6F:B8:05:A3:75:A1:A4:1C:6B:1E:55:02:A4:F9:6B:CA:F7:E6"))
    .attribute("setup", Some("actpass"))
    .attribute("mid", Some("1"))
    .attribute("rtpmap", Some("96 VP8/90000"))
    .attribute("rtcp-fb", Some("96 ccm fir"))
    .attribute("rtcp-fb", Some("96 nack"))
    .attribute("rtcp-fb", Some("96 nack pli"))
    .attribute("rtcp-fb", Some("96 goog-remb"))
    .attribute("rtpmap", Some("97 rtx/90000"))
    .attribute("fmtp", Some("97 apt=96"))
    .attribute("rtpmap", Some("98 H264/90000"))
    .attribute("rtcp-fb", Some("98 ccm fir"))
    .attribute("rtcp-fb", Some("98 nack"))
    .attribute("rtcp-fb", Some("98 nack pli"))
    .attribute("fmtp", Some("98 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f"))
    .attribute("ssrc-group", Some("FID 1629250457 3327211740"))
    .attribute("ssrc", Some("1629250457 cname:ConferenceTool"))
    .attribute("ssrc", Some("1629250457 msid:myWebRTCStream videoTrack"))
    .attribute("ssrc", Some("1629250457 mslabel:myWebRTCStream"))
    .attribute("ssrc", Some("1629250457 label:videoTrack"))
    .done()
    .build()
    .unwrap();

// Create a WebRTC SIP INVITE
let request = SimpleRequestBuilder::new(Method::Invite, "sip:webrtc-gateway.example.com").unwrap()
    .from("WebRTC User", "sip:webuser@example.com", Some("webrtc1"))
    .to("SIP Peer", "sip:sipuser@example.com", None)
    .contact("<sip:webuser@198.51.100.1:5060;transport=ws>", None)
    .sdp_body(&sdp)  // Sets Content-Type to application/sdp automatically
    .build();

Required Methods§

Source

fn sdp_body(self, sdp: &SdpSession) -> Self

Add an SDP session as the message body

This method sets the Content-Type to application/sdp and adds the SDP session as the message body. It automatically handles setting both the header and the body.

SDP (Session Description Protocol) is defined in RFC 4566 and is commonly used in SIP for media negotiation, particularly in INVITE requests and their responses.

§Parameters

sdp: The SDP session to add as the body

§Returns

Self for method chaining

§Examples

§Basic Audio Call Setup

use rvoip_sip_core::builder::SimpleRequestBuilder;
use rvoip_sip_core::builder::headers::ContentBuilderExt;
use rvoip_sip_core::types::Method;
use rvoip_sip_core::sdp::SdpBuilder;

// Create an SDP builder for a basic audio call
let sdp = SdpBuilder::new("Phone Call")
    .origin("alice", "123", "456", "IN", "IP4", "192.0.2.1")
    .connection("IN", "IP4", "192.0.2.1")
    .time("0", "0")
    .media("audio", 49170, "RTP/AVP")
    .formats(&["0", "8"])  // PCMU and PCMA
    .attribute("rtpmap", Some("0 PCMU/8000"))
    .attribute("rtpmap", Some("8 PCMA/8000"))
    .attribute("ptime", Some("20"))
    .done()
    .build()
    .unwrap();

// Create an INVITE request with the SDP body
let invite = SimpleRequestBuilder::invite("sip:bob@example.com").unwrap()
    .from("Alice", "sip:alice@example.com", Some("tag123"))
    .to("Bob", "sip:bob@example.com", None)
    .contact("<sip:alice@192.0.2.1:5060>", None)
    .sdp_body(&sdp)
    .build();

§SIP Response with SDP Answer

use rvoip_sip_core::builder::SimpleResponseBuilder;
use rvoip_sip_core::builder::headers::ContentBuilderExt;
use rvoip_sip_core::types::StatusCode;
use rvoip_sip_core::sdp::SdpBuilder;

// Create an SDP answer selecting specific codecs
let sdp = SdpBuilder::new("Answer")
    .origin("bob", "789", "012", "IN", "IP4", "192.0.2.2")
    .connection("IN", "IP4", "192.0.2.2")
    .time("0", "0")
    .media("audio", 45678, "RTP/AVP")
    .formats(&["0"])  // Only accepting PCMU
    .attribute("rtpmap", Some("0 PCMU/8000"))
    .attribute("ptime", Some("20"))
    .done()
    .build()
    .unwrap();

// Create a 200 OK response with the SDP answer
let response = SimpleResponseBuilder::new(StatusCode::Ok, Some("OK"))
    .from("Bob", "sip:bob@example.com", Some("tag456"))
    .to("Alice", "sip:alice@example.com", Some("tag123"))  // Preserve tag from request
    .contact("<sip:bob@192.0.2.2:5060>", None)
    .sdp_body(&sdp)
    .build();

§Conference Call Setup

use rvoip_sip_core::builder::SimpleRequestBuilder;
use rvoip_sip_core::builder::headers::ContentBuilderExt;
use rvoip_sip_core::types::Method;
use rvoip_sip_core::sdp::SdpBuilder;

// Create SDP for a conference call with multiple audio formats
let sdp = SdpBuilder::new("Conference Call")
    .origin("confserver", "2890844527", "2890844527", "IN", "IP4", "conference.example.com")
    .info("Conference call with audio mixing")
    .connection("IN", "IP4", "conference.example.com")
    .time("0", "0")
    .media("audio", 49170, "RTP/AVP")
    .formats(&["0", "8", "96", "97"])
    .attribute("rtpmap", Some("0 PCMU/8000"))
    .attribute("rtpmap", Some("8 PCMA/8000"))
    .attribute("rtpmap", Some("96 opus/48000/2"))
    .attribute("rtpmap", Some("97 telephone-event/8000"))
    .attribute("fmtp", Some("97 0-16"))
    .attribute("ssrc", Some("1234567890 cname:conf@example.com"))
    .attribute("ssrc", Some("1234567890 label:conference-mixer"))
    .attribute("maxptime", Some("40"))
    .attribute("recvonly", None::<&str>)
    .done()
    .build()
    .unwrap();

// Create an INVITE to join a conference
let invite = SimpleRequestBuilder::invite("sip:conference@example.com").unwrap()
    .from("Alice", "sip:alice@example.com", Some("confcall"))
    .to("Conference", "sip:conference@example.com", None)
    .contact("<sip:alice@192.0.2.1:5060>", None)
    .sdp_body(&sdp)
    .build();

Dyn Compatibility§

This trait is not dyn compatible.

In older versions of Rust, dyn compatibility was called "object safety", so this trait is not object safe.

Implementors§

Source §

impl ContentBuilderExt for SimpleRequestBuilder

Implementation for new SimpleRequestBuilder

Source §

impl ContentBuilderExt for SimpleResponseBuilder

Implementation for new SimpleResponseBuilder

Trait ContentBuilderExtCopy item path

§SIP Message Content Overview

§Purpose of Message Content

§Common Content Types in SIP

§Relationship with other headers

§Examples

§INVITE with SDP for Audio Call

§Audio and Video Call with SRTP

§WebRTC Media Negotiation

Required Methods§

fn sdp_body(self, sdp: &SdpSession) -> Self

§Parameters

§Returns

§Examples

§Basic Audio Call Setup

§SIP Response with SDP Answer

§Conference Call Setup

Dyn Compatibility§

Implementors§

impl ContentBuilderExt for SimpleRequestBuilder

impl ContentBuilderExt for SimpleResponseBuilder

Trait ContentBuilderExt