Interview Questions.

Question 1. Is Rtp A Transport Protocol Or A Kind Of Application Protocol?

Answer :

RTP has critical residences of a transport protocol:

it runs on give up structures, it provides demultiplexing. It differs from delivery protocols like TCP in that it (currently) does not offer any shape of reliability or a protocol-described waft/congestion manage. However, it offers the important hooks for adding reliability, where appropriate, and float/congestion manage. Some like to consult this belongings as software-level framing (see D. Clark and D. Tennenhouse, "Architectural concerns for a brand new era of protocols", SIGCOMM'90, Philadelphia). RTP so far has been broadly speaking implemented within packages, however that has no referring to its position. TCP is still a shipping protocol even if it is implemented as a part of an utility rather than the working machine kernel.

Question 2. Rtp Does Not Ensure Real-time Delivery. So How Come It Is Called A Real-time Protocol?

Answer :

No stop-to-stop protocol, which include RTP, can make sure in-time delivery. This usually calls for the assist of decrease layers that truly have manage over resources in switches and routers. RTP offers capability applicable for sporting real-time content, e.G., a timestamp and manage mechanisms for synchronizing distinctive streams with timing houses.

Voice Over Internet Protocol (VOIP) Interview Questions
Question 3. Is Rtp An Unreliable Protocol? Are There Any Mechanisms Provided For Error Recovery In Rtp?

Answer :

As currently defined, RTP does no longer define any mechanisms for convalescing for packet loss. Such mechanisms are probable to be quite dependent on the packet content material. For instance, for audio, it has been cautioned to feature low-bit-rate redundancy, offset in time. For other programs, retransmission of lost packets may be suitable. (The H.261 RTP payload definition gives one of these mechanism.) This requires no additions to RTP. RTP likely has the essential header information (like sequence numbers) for a few varieties of blunders healing by retransmission.

Question four. Can Rtp Run Over Ipv6? Atm?

Answer :

Yes. RTP includes no precise assumptions approximately the skills of the decrease layers, except that they provide framing. It carries no community-layer addresses, so that RTP is not suffering from addressing modifications. Any extra decrease-layer skills such as safety or great-of-provider guarantees can glaringly be utilized by programs using RTP. There are several implementations of video tools that run RTP without delay over AAL5 (T. Braun) and recent efforts to outline the carriage of RTP over AAL2 and AAL5. It must be referred to that the RTCP CNAME area is currently based totally on the belief that hosts have Internet-style domain names.

Internet Protocol model 6 (IPv6) Tutorial
Question five. Can Rtp Be Used In Asymmetric Networks?

Answer :

In asymmetric networks, the bandwidth in one path, commonly from the user to the Internet, is extensively lower than inside the different. These networks encompass ADSL, cable modems and satellite distribution. RTP may be used easily, but it may be important to have most effective information senders send RTCP messages. These RTCP messages are beneficial to permit inter-media synchronization and discover the content material of the media stream.

Veritas Volume Manager (VVM or VxVM) Interview Questions
Question 6. Why Doesn't Rtp Have A Length Field?

Answer :

RTP does no longer comprise a period discipline, that is, it assumes that framing is executed by using the underlying protocol and that only one RTP packet is to be carried in a single PDU of the underlying protocol. This is the everyday application with UDP (or AAL5) because the underlying protocol. Since most programs currently predicted do not want framing, it would be a waste of processing and bandwidth to feature one. This is protected in detail inside the section RTP over Network and Transport Protocols of the spec.

If RTP is used with a protocol that isn't message-based (e.G., TCP) or if it is proper to hold several RTP packets in a single lower-layer PDU (e.G., for aggregation of streams), it's far trivial to define a profile that prefixes the RTP header by means of a 16 or 32-bit duration field, relying at the preferred tradeoff among overhead and keeping word alignment.

Question 7. Does Rtp Have A Fixed Packetization Interval?

Answer :

Some implementations count on that packet audio is sent with a selected packetization c programming language, e.G., 20 ms. This is inaccurate. While RFC 1890 recommends certain values and SDP permits to specific a preference, implementations want on the way to cope with all affordable values. There is no constraint that G.711 or different sample-based codecs is conveyed in multiples of a certain unit. Thus, an RTP packet with 123 samples of G.711 is perfectly valid and needs to be dealt with accurately.

Internet Protocol version four (IPv4) Tutorial Simple Mail Transfer Protocol (SMTP) Interview Questions
Question 8. How Does Padding Work?

Answer :

Since the underlying delivery unit defines the give up of the packet, the application can constantly locate the ultimate byte of the (say, UDP) packet and appearance there for the range of padding bytes.

Question nine. Practically Speaking, How Is The Timestamp Computed?

Answer :

For audio, the timestamp is incremented by using the packetization c programming language instances the sampling fee. For example, for audio packets containing 20 ms of audio sampled at eight,000 Hz, the timestamp for each block of audio will increase through 160, even if the block is not sent due to silence suppression. Also, note that the real sampling charge will differ slightly from this nominal charge, but the sender usually has no dependable way to measure this divergence.

For video, time clock charge is fixed at ninety kHz. The timestamps generated depend on whether the application can decide the frame range or no longer. If it can or it could make sure that it's miles transmitting every body with a hard and fast body rate, the timestamp is ruled by way of the nominal frame price. Thus, for a 30 f/s video, timestamps might boom by means of three,000 for each frame, for a 25 f/s video by 3,six hundred for each body. If a frame is transmitted as several RTP packets, these packets could all endure the equal timestamp. If the frame number can't be decided or if frames are sampled aperiodically, as is normally the case for software codecs, the timestamp needs to be computed from the machine clock (e.G., gettimeofday()).

Internet Protocol version 6 (IPv6) Interview Questions
Question 10. In A Multimedia Conference, Are The Initial Timestamp Values Related?

Answer :

No, initial time stamp values are picked randomly and independently for each RTP circulation. (This is more or much less unavoidable if exclusive media sorts are generated by using independent programs, whether or not these applications live at the same host or now not.) Synchronization (along with lip sync) between one of a kind media is executed by using receivers thru the NTP timestamps in the RTCP sender reports. This timestamp affords a not unusual time reference that associates a media-particular RTP timestamp with the commonplace "wallclock" time shared throughout media. The mechanism how end structures synchronize one of a kind media is not prescribed by way of RTP, however, a manageable method is to periodically trade messages between programs to suggest what put off every application might impose on the flow (together with any media decoding delays) if it were not to synchronize after which have all applications select the most of these delays.

Question eleven. What Are The Roles Of The Rtp Timestamp And Sequence Numbers?

Answer :

The timestamp is used to place the incoming audio and video packets in the correct timing order (playout delay compensation). The collection quantity is particularly used to stumble on losses. Sequence numbers increase by one for every RTP packet transmitted, timestamps increase by the point "included" with the aid of a packet. For video codecs in which a video body is cut up throughout numerous RTP packets, numerous packets may have the identical timestamp. In a few cases including carrying DTMF (contact tone) statistics (RFC 2833), RTP timestamps may not be monotonic.

Internet Protocol version four (IPv4) Interview Questions
Question 12. What Are The Different Clocks And How Are They Synchronized?

Answer :

RFC 3550 specifies one media-timestamp within the RTP facts header and a mapping between such timestamp and a globally synchronized clock, carried as RTCP timestamp mappings.

The NTP timestamps within the SR are assumed to be synchronized between all media senders within a unmarried consultation. If the media resources come from the identical community source, that is obviously now not an trouble. Receiver(s) synchronize to the sender, the only solution possible for multicast.

Experience has shown that every one other cross-media, pass-host schemes come to be doing clock synchronization, generally not so good as NTP and application-specific.

Voice Over Internet Protocol (VOIP) Interview Questions
Question 13. What's The Marker Bit Good For?

Answer :

For voice packets, the marker bits indicates the beginning of a talkspurt. Beginning of talkspurts are true possibilities to modify the playout postpone at the receiver to compensate for variations among the sender and receiver clock rates in addition to modifications inside the community put off jitter. Packets in the course of a talkspurt want to played out constantly, even as listeners generally are not touchy to moderate variations in the durations of a pause.

The marker bit is a hint; the beginning of a talkspurt also can be computed through comparing the distinction in timestamps and sequence numbers among  packets, assuming the timestamp clock charge is thought.

Question 14. What Is The Sender Packet Count And Byte Count Used For?

Answer :

They are not needed for loss computation; the sequence quantity fields are used for that to keep away from spherical-off mistakes. They may be used to compute the sender packet and byte fee.

Question 15. What Is The Rtp Timestamp In The Rtcp Sender Report Used For?

Answer :

The RTP timestamp and NTP timestamps shape a couple that perceive absolutely the time of a specific pattern within the circulate. For example, if the RTCP sender document includes an RTP timestamp of 1234 and an NTP timestamp indicating February 3, 10:14:15, it manner that sample 1234 inside the media circulation occured exactly on February 3, 10:14:15.

Spanning Tree Protocol (STP) Interview Questions
Question 16. How Is The Jitter Computed?

Answer :

If numerous packets, say, inside a video frame, endure the same timestamp, it's miles really helpful to only use the primary packet in a frame to compute the jitter. (This problem can be addressed in a destiny model of the specification.)

Jitter is computed in timestamp units. For instance, for an audio move sampled at 8,000 Hz, the arrival time measured with the neighborhood clock is transformed through multiplying the seconds with the aid of eight,000.

Steve Casner wrote:

For encodings such as MPEG that transmit statistics in a special order than it become sampled, this provides noise into the jitter calculation. I actually have heard handwavy arguments that this factor may be calculated out given which you recognise the form of the noise, but my math isn't always robust enough for that.

In a number of the cases that we care about, the jitter added through MPEG might be small sufficient that once the network jitter is of the identical order we don't have a hassle anyway.

There is any other trouble for video in that each one of the packets of a frame have the same timestamp because the whole frame is sampled at once. However, the dispersion in time of these packets genuinely is all a part of the community transfer manner that the receiver should accommodate with its buffer.

It has been cautioned that jitter be calculated handiest on the primary packet of a video body, or simplest on "I" frames for MPEG. However, that can colour the consequences also due to the fact the ones packets may additionally see transit delays distinct than the subsequent packets see.

The major point to keep in mind is that the number one function of the RTP timestamp is to represent the inherent belief of real time related to the media. It additionally turns out to be useful for the jitter degree, however that is a secondary characteristic.

The jitter fee isn't always predicted to be beneficial as an absolute fee. It is more useful as a method of evaluating the reception nice at two receiver or comparing the reception satisfactory five mins ago to now

Question 17. What Is The Session Bandwidth?

Answer :

First, it's miles most surely now not the link bandwidth. This could now not scale, as then a large variety of periods ought to saturate the hyperlink with RTCP traffic, even if each used just 5% of the link bandwidth for RTCP. Secondly, the concept of hyperlink bandwidth is ill-described in a heterogeneous network.

The session bandwidth is the nominal records bandwidth plus the IP, UDP and RTP headers (forty bytes). For instance, for sixty four kb/s PCM audio packetized in 20 ms increments, the session bandwidth might be (a hundred and sixty + 40) / zero.02 bytes/2nd or 80 kb/s. If there are more than one senders, the sum in their person bandwidths is used.

The consultation bandwidth is usually described out-of-band, e.G., in a consultation assertion protocol, based on affordable estimates of the quantity of concurrent senders and their common bandwidth. Distributed and constant on-line estimation of the consultation bandwidth can be tough as the number of senders and their bandwidth adjustments. The absolute fee is less vital than that all individuals agree on a common value. (After all, there's nothing unique about choosing the RTCP bandwidth to be 5% of the consultation bandwidth, it simply needs to be agreed upon by way of all individuals to avoid timing out members in advance.)

Border Gateway Protocol (BGP) Interview Questions
Question 18. What Is The Use Of Rtcp For Two-party Calls?

Answer :

Since the price of sending RTCP is minimal (approximately one packet each 5 seconds), it makes feel to ship RTCP even for factor-to-factor connections:

With RTCP, both sides recognize how well the opposite side is receiving audio and video; that is useful, since degraded great could have any quantity of motives past community loss, delay and jitter. A particular use is whilst calling technical guide: the tech support character can look at the network overall performance on the far flung stop.

RTCP is essential for synchronizing audio and video streams.
For audio with silence suppression, RTCP is beneficial as a liveness indication.
The SDES statistics is beneficial for user interfaces.
Many programs will (want to) aid both unicast and multicast, in order that the additional implementation complexity is zero.
Veritas Volume Manager (VVM or VxVM) Interview Questions
Question 19. How Do I Register An Rtp Payload Type?

Answer :

See the outline, drawn from RFC 1890 (with some realistic comments).

Question 20. What Are Dynamic Payload Types?

Answer :

Dynamic payload types are described inside the RTP A/V Profile. Unlike static payload kinds, dynamic payload types aren't assigned in the RTP A/V Profile or by using IANA. They map an RTP payload type to an audio and video encoding throughout a consultation. Different members of a consultation ought to, however normally do not, use one of a kind mappings. Dynamic payload sorts use the variety 96 to 127. They are assigned through way outdoor of the RTP profile or protocol specification, along with

consultation descriptions like SDP (the usage of the a:rtpmap parameter), used in bulletins and invitations (e.G., SIP);

for instance:

  m=audio 12345 RTP/AVP/121

  a=rtpmap:121 RT24

other signaling protocols (however, H.245 does now not appear to have a mechanism for doing this, as a minimum no longer for non-ITU protocols).

Note that a number of encodings are defined inside the RTP A/V profile which do now not have a static (permanent) payload type. The RTP A/V Profile defines names for encodings which can be used by SDP or other mechanisms to specify the mapping. Encodings may also be diagnosed by object identifiers or other names.

Since the distance for payload sorts is restricted, simplest very common encodings need to be assigned static sorts. These are generally audio and video encodings "blessed" by using global standardization bodies, including the G. Series of ITU-T audio encodings. The RTP A/V Profile defines a hard and fast of criteria for making static assignments.

Enhanced Interior Gateway Routing Protocol (EIGRP) Interview Questions
Question 21. If I'm Using H.323 Or Other Set-up Protocol, Can I Ignore The Rtp Payload Type (pt) Field?

Answer :

An application must by no means simply play a packet with out examining its payload type, even if a single payload kind has been negotiated thru H.245 or similar protocols. New mechanisms, along with transmission of DTMF digits (RFC 2833), consolation noise indication, ahead error correction the use of redundant statistics, switching of encodings to bear in mind community conditions may also simply use the PT to suggest special packets, which an quit software can ignore, if preferred, ensuring backward compatibility. But this assumption is violated if an application blindly performs returned all packets irrespective of PT. Also, in multicast environments, it's far not likely that each sender will use the equal payload type.

Question 22. Should The Rtp Payload Type (pt) Field Be Used For Multiplexing Different Streams?

Answer :

It has been recommended that in some environments (which include RTP over AAL5) that lack lower-layer muxing skills, the RTP payload type (PT) subject be used to distinguish streams originating from extraordinary assets. This is a basically terrible concept and violates the letter and intent of the specification. It makes use of multiple PTs in a unmarried circulate tough (see preceding question). It is likewise unnecessary, as the SSRC turned into designed for distinguishing several sources.

Question 23. Should The Rtp Ssrc Be Used For Demultiplexing Different Streams For The Same Rtp Session?

Answer :

The RTP SSRC is meant to label streams from unique resources, this is, every sender in a convention has its personal SSRC. It has been recommended to have a unmarried supply, the usage of the identical RTP consultation (diagnosed by source and vacation spot addresses and ports), send special media, which include an audio and video movement, the use of special SSRCs.

This is typically a terrible idea for the subsequent motives:

An RTP mixer commonly combines all of the SSRCs it receives on an RTP consultation in keeping with the composition method this is suitable for that consultation (e.G., mixing for audio). If a couple of media are sent on one consultation, then the SSRCs need to be segregated in keeping with medium based on external statistics. That receives complicated with assets coming from more than one places. It is in addition more complex for and end node receiver to handle streams coming from more than one assets to the equal RTP session if a number of the ones assets don't all get fed to the identical compositor (mixer, selector, something).

Carrying multiple media in a single RTP session precludes the use of different network paths or community aid allocations if appropriate. For the everyday synchronized audio/video move one won't need exceptional paths, however it isn't always hard to imagine conditions where one medium ought to cross through a low-bandwidth, low-postpone terrestrial direction while another can tolerate the longer delay of a satellite route with a purpose to get higher bandwidth.

Carrying more than one media in one RTP consultation precludes reception of a subset of the media if preferred, as an example just audio if video might exceed the available bandwidth. This is not an problem for unicast since that desire of media could be controlled by using the alternate with the sender, but it is precious for multicast with heterogeneous receivers.

Carrying multiple media in one RTP session precludes receiver implementations that use separate methods for the different media, while the usage of separate RTP sessions allows either single- or multiple-system implementations. Consider the development of "table location networks" at MIT, ISI and different places in which the show and the speaker may have specific IP addresses. This is an example of the overall philosophy of demultiplexing at the lowest stage feasible.

Also, making the SSRC fixed is a hassle inside the multicast case due to the fact collision resolution might require converting the SSRC identity.(contributed via Steve Casner)

Post Office Protocol (POP) Interview Questions
Question 24. Do Receivers Need Their Own Ssrc Identifiers?

Answer :

Yes, all contributors in an RTP session have SSRC values, considering the fact that they're needed in receiver reviews.

Simple Mail Transfer Protocol (SMTP) Interview Questions
Question 25. Why Can't We Just Use Tcp For Audio And Video?

Answer :

For handing over audio and video for playback, TCP may be suitable. Also, with sufficiently lengthy buffering and adequate average throughput, near-real-time shipping the use of TCP can be a success, as practiced via the Netscape WWW browser. TCP may also frequently run over noticeably lossy networks (e.G., the German X.25 network) with applicable throughput, even though the uncompensated losses could make audio or video communique impossible.

However, for real-time transport of audio and video, TCP and other reliable delivery protocols which include XTP are irrelevant. The three main reasons are:

Reliable transmission is irrelevant for put off-sensitive information such as actual-time audio and video. By the time the sender has located the lacking packet and retransmitted it, at the least one round-journey time, probable greater, has elapsed. The receiver either has to anticipate the retransmission, increasing postpone and incurring an audible gap in playout, or discard the retransmitted packet, defeating the TCP mechanism. Standard TCP implementations force the receiver software to attend, in order that packet losses would constantly yield increased delay. Note that a unmarried packet lost time and again may want to substantially boom postpone, which would persist at the least till the stop of talkspurt.

TCP can not assist multicast.

The TCP congestion control mechanisms decreases the congestion window when packet losses are detected ("gradual begin"). Audio and video, then again, have "herbal" fees that can't be  decreased without starving the receiver. For example, standard PCM audio calls for sixty four kb/s, plus any header overhead, and can't be introduced in much less than that. Video will be more effortlessly throttled sincerely via slowing the acquisition of frames at the sender whilst the transmitter's send buffer is complete, with the corresponding put off. The accurate congestion response for these media is to trade the audio/video encoding, video body rate, or video photo size at the transmitter, based totally, for example, on feedback acquired thru RTCP receiver document packets.

An additional small disadvantage is that the TCP and XTP headers are large than a UDP header (forty bytes for TCP and XTP 3.6, 32 bytes for XTP 4.0, as compared to eight bytes). Also, those dependable delivery protocols do now not comprise the vital timestamp and encoding facts wished by way of the receiving application, so that they can not replace RTP. (They could not need the sequence range as those protocols guarantee that no losses or reordering takes place.)

While LANs often have sufficient bandwidth and coffee sufficient losses no longer to cause these troubles, TCP does not provide any blessings in that state of affairs either, except for the healing from rare packet losses. Even in a LAN with out a losses, the TCP gradual begin mechanism could restrict the initial charge of the supply for the first few spherical-journey times.

Question 26. Can't We Just Use Xtp?

Answer :

Many of the arguments parallel those inside the previous section. The question of the relationship of RTP and XTP appears to rise up frequently. (This might also clearly be because of the phrase 'transport' in each protocol names.) However, XTP and RTP are not replacements for every different. XTP is designed as a trendy, configurable network and shipping protocol for each reliable and unreliable facts communications. RTP has no reliability mechanisms (even though these will be added if desired for specific applications) and no go with the flow manipulate just like the price control in XTP. RTP isn't always intended for normal, dependable records transfer (where TCP or XTP might be used rather). For actual-time data, in which retransmission is typically not possible due to timing constraints, XTP would ought to disable retransmission. Flow/congestion manage for actual-time information is most likely irrelevant because the price of such resources is inherently given and now not modifiable at the time-scale of delivery-protocol flow manage, as explained inside the previous section. It must be mentioned that RTP supports mechanisms that permit a form of congestion manage on longer time scales, e.G., by way of modifying the supply encoder if network congestion is detected.

RTP has no protocol nation by itself and might as a result be used over either connection-much less networks, inclusive of IP/UDP, or connection-oriented networks, which include XTP, ST-II or ATM (AAL3/4 or AAL5). Many real-time multimedia programs use multicast with a massive fan-out, e.G., numerous hundred to hundreds for a lecture or concert. Connection-orientated protocols like XTP have trouble scaling to one of these massive quantity of receivers.

XTP does no longer provide timing or content material kind (media) records, and for that reason could need these services, as provided by means of RTP. XTP offers no RTP-like direct remarks of the acquired satisfactory-of-provider, and hence, once more, would ought to "import" these from any other protocol. Looking at current applications using XTP for actual-time services confirms that they want to feature a layer comparable in content material to the RTP facts part "between" XTP and the actual media.

Question 27. How Should Rtp Sessions Be Played Back?

Answer :

Since RTCP packets incorporate absolute time information, a recorded consultation can not honestly be performed lower back by way of time-moving the entire recorded consultation. One approach performs again the statistics packets with their unique time stamps, with re-normalized timing. SDES information apart from NOTE objects can be accumulated for each source and regenerated as in a "live" session. NOTE SDES objects need to be inserted at the right on the spot in the playback as they're allowed to exchange.

Internet Protocol model 6 (IPv6) Interview Questions
Question 28. What Are Some Of The Differences Between The Vat Protocol And Rtp?

Answer :

The VAT protocol was at first carried out within the VAT audio tool and sooner or later also in different audio equipment including NeVoT. The VAT protocol is now out of date and must not be used or applied.

The VAT header layout is best described in header files. (See the VAT and NeVoT resources for info.) Many factors of RTP and the VAT protocol are comparable, but RTP improves upon the VAT protocol in a number of methods:

The VAT protocol changed into designed for audio handiest, at the same time as RTP is certain for audio and video and may be suitable for other real-time packages.

RTP is designed to be protocol-independent and may be used with non-IP protocols (ATM AAL5, as an example) as well as, say, IPv6.

RTP supply identity simplifies the usage of mixers and translators.

RTP has some of features that simplify use of software-degree encryption (padding, and many others.).

The RTP header is extensible, should the want rise up within the future.

The RTP header has a sequence number which simplifies accurate loss detection and size and the dealing with of pics transmitted in several packets.

The RTCP SDES packets include additional records that simplify tracing of misbehaving assets, e.G., their e mail deal with or cellphone range.

The RTCP SDES CNAME gadgets simplify the development of multimedia software from independent media agents.

RTCP sender and receiver reviews permit the implementation of adaptive programs, that is, programs wherein senders scale their bandwidth consumption primarily based on network load.

RTCP sender and receiver reviews permit tracking of the nice of carrier inside, say, a multimedia convention.

Question 29. What Are The Differences Between Rtp Version 1 And 2?

Answer :

Version 1 is of ancient hobby only. Applications need to no longer be written for it. RTP model 2 isn't backwards compatible with version 1. If you care, you may find a definition of model 1 in an vintage Internet draft.

Question 30. What About Firewalls?

Answer :

Ports used:

H.323 TCP 1720

H.235 TCP ephemeral, > 1024

Question 31. What Is The Quality Of Audio Codec X?

Answer :

See separate summary with audio samples.

Question 32. Are All Audio Codecs Patented?

Answer :

Most older, better-bitrate codecs aren't problem to patent safety. However, G.723, G.729.1 and GSM are protected via diverse patents. For example, U.S. Patent four,752,956, Digital speech coder with baseband residual coding modifies coding using short time period pleasant shape speech statistics produced with the aid of analysers inside encoder-multiplexers applies to GSM and is assigned to Philips.

Question 33. Are There Other Efforts In Using The Internet For Real-time Audio And Video?

Answer :

Too many, some may additionally say. Vat versions 3.Four and in advance, one of the early (latest) Internet audio programs, uses primarily the equal audio encodings as unique in the RTP profile, however a specific protocol. There also are some of Internet telephony programs that typically most effective operate on PCs and in unicast mode. There are initial efforts to interconnect the public switched phone community and the Internet.

CuSeeMe (for Windows PC and the Macintosh) is a combined audio and video tool using reflectors instead of IP-degree multicast.

The Internet Telephony Consortium continues a list of standards and associated efforts.

Internet Protocol model 4 (IPv4) Interview Questions
Question 34. Is There An Rtp Library Or Kernel Implementation?

Answer :

RTP (particularly, the statistics component) is tightly coupled to the utility, so that a kernel implementation makes little experience. A range of humans have advanced libraries that put in force RTP and RTCP (see list). The sources to NeVoT, rtpdump, vat, rat and vic additionally incorporate RTP and RTCP processing modules which ought to be usable in different packages with minor adjustments. Note also that the specification itself carries severa code fragments. (Most of the opposite programs are the usage of older versions of RTP and hence should not be relied upon for developments.)

The Java Media Framework (JMF), a Java API, also helps RTP and RTCP.

There is no widespread API for RTP.