RTP does not have a standard TCP or UDP port on which it communicates. The only standard that it obeys is that UDP communications are done via an even port and the next higher odd port is used for RTP Control Protocol (RTCP) communications. Although there are no standards assigned, RTP is generally configured to use ports 16384-32767. RTP can carry any data with real-time characteristics, such as interactive audio and video. Call setup and tear-down for VoIP applications is usually performed by either SIP or H.323 protocols. The fact that RTP uses a dynamic port range makes it difficult for it to traverse firewalls. In order to get around this problem, it is often necessary to set up a STUN server.
According to RFC 1889, the services provided by RTP include:
The protocols themselves do not provide mechanisms to ensure timely delivery. They also do not give any Quality of Service (QoS) guarantees. These things have to be provided by some other mechanism.
Also, out of order delivery is still possible, and flow and congestion control are not supported directly. However, the protocols do deliver the necessary data to the application to make sure it can put the received packets in the correct order. Also, RTCP provides information about reception quality which the application can use to make local adjustments. For example if a congestion is forming, the application could decide to lower the data rate.
RTP was also published by the ITU-T as H.225.0, but later removed once the IETF had a stable standards-track RFC published. It exists as an Internet Standard (STD 64) defined in RFC 3550 (which obsoletes RFC 1889). RFC 3551 (STD 65) (which obsoletes RFC 1890) defines a specific profile for Audio and Video Conferences with Minimal Control. RFC 3711 defines the Secure Real-time Transport Protocol (SRTP) profile (actually an extension to RTP Profile for Audio and Video Conferences) which can be used (optionally) to provide confidentiality, message authentication, and replay protection for audio and video streams being delivered.
The RTP header size is minimum 12 bytes. Ver.: (2 bits) Indicates the version of the protocol. Current version is 2. P : (1 bit) Used to indicate if there are extra padding bytes at the end of the RTP packet. X : (1 bit) Indicates presence of an Extension header between standard header and payload data. CC : (4 bits) Contains the number of CSRC identifiers that follow the fixed header. M : (1 bit) Used at the application level and is defined by a profile. If it is set, it means that the current data has some special relevance for the application. PT : (7 bits) Indicates the format of the payload and determines its interpretation by the application. Sequence Number : (16 bits) The sequence number increments by one for each RTP data packet sent, and may be used by the receiver to detect packet loss and to restore packet sequence. Timestamp : (32 bits) The timestamp reflects the sampling instant of the first data in the RTP data packet. The sampling instant MUST be derived from a clock that increments monotonically and linearly in time to allow synchronization and jitter calculations. The resolution of the clock MUST be sufficient for the desired synchronization accuracy and for measuring packet arrival jitter (one tick per video frame is typically not sufficient). The clock frequency is dependent on the format of data carried as payload and is specified statically in the profile or payload format specification that defines the format, or MAY be specified dynamically for payload formats defined through non-RTP means. If RTP packets are generated periodically, the nominal sampling instant as determined from the sampling clock is to be used, not a reading of the system clock. SSRC : Synchronization source identifier uniquely identifies the source of a stream. CSRC : Contributing source IDs enumerate contributing sources to a stream which has been generated from multiple sources. Extension header : The first 32-bit word contains a profile specific identifier (16 bits) and a length specifier (16 bits) that indicates the length of the extension (EHL=extension header length) in 32-bit units, excluding the 32 bits of the extension header.
The Real-time Transport Protocol (RTP) and the Real-time Transport Control Protocol (RTCP) are commonly used together. RTP is used to transmit data (e.g. audio and video) and RTCP is used to monitor QoS.
To reduce the size of the IP, UDP and RTP headers, Compressed RTP (CRTP) was developed and specified in RFC 2509. It is primarily used for reliable and fast point-to-point links, but it can be problematic in other applications. Therefore, Enhanced CRTP (ECRTP) was defined.
Especially in VoIP over wireless applications, headers are significantly larger than the payload. The Robust Header Compression (ROHC) specified in RFC 3095 seems to be an increasingly deployed method for better efficiency. There are currently two ROHC profiles defined for the compression of IP/UDP/RTP traffic. The original definition in RFC 3095, and a recently published RFC 5225.