Rfc 4733 txt

Notify me of new posts via email. Email Address:. Tao, Zen, and Tomorrow. About Andrew. The final packet sets End of Event to True 1 and will look like this: At this point, the tone has stopped playing and the application can process it as it sees fit.

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Late Media. Create a free website or blog at WordPress. Follow Following. Tao, Zen, and Tomorrow Join 2, other followers. Sign me up. Already have a WordPress. Log in now. If the sender receives an "events" parameter from the receiver, it MUST restrict the set of events it sends to those listed in the received "events" parameter.

Section 2. The following SDP shows an example of such usage, where G. This value does not have to be the same in both directions. The appropriate period may vary with the application, since increased packetization periods imply increased end-to-end response times in instances where one end responds to events reported from the other. Negotiation of telephone-events sessions using SDP MAY specify such differences by separating events corresponding to different applications into different streams.

In the example below, events are DTMF events, which have a fairly wide tolerance on timing. Events and are events related to data transmission and are subject to end-to-end response time considerations. As a result, they are assigned a smaller packetization period than the DTMF events. Transmission of Event Packets DTMF digits and other named telephone events are carried as part of the audio stream, and they MUST use the same sequence number and timestamp base as the regular audio channel to simplify the generation of audio waveforms at a gateway.

The update packets MUST have the same RTP timestamp value as the initial packet for the event, but the duration MUST be increased to reflect the total cumulative duration since the beginning of the event. Sending of a packet with the E bit set is OPTIONAL if the packet reports two events that are defined as mutually exclusive states, or if the final packet for one state is immediately followed by a packet reporting a mutually exclusive state.

For events defined as states, the appearance of a mutually exclusive state implies the end of the previous state. A source has wide latitude as to how often it sends event updates. A natural interval is the spacing between non-event audio packets. Recall that a single RTP packet can contain multiple audio frames for frame-based codecs and that the packet interval can vary during a session.

Timing information is contained in the RTP timestamp, allowing precise recovery of inter-event times. Thus, the sender does not in theory need to maintain precise or consistent time intervals between event packets. However, the sender SHOULD minimize the need for buffering at the receiving end by sending event reports at constant intervals. DTMF digits and other tone events are sent incrementally to avoid having the receiver wait for the completion of the event.

In some cases for example, data session startup protocols , waiting until the end of a tone before reporting it will cause the session to fail. In other cases, it will simply cause undesirable delays in playout at the receiving end.

The sender MUST then begin reporting a new "segment" with the RTP timestamp set to the time at which the previous segment ended and the duration set to the cumulative duration of the new segment. The sender MUST repeat this procedure as required until the end of the complete event has been reached. The final packet for the complete event MUST have the E bit set either on initial transmission or on retransmission as described below. Exceptional Procedure for Combined Payloads If events are combined as a redundant payload with another payload type using RFC [ 2 ] redundancy, the above procedure SHALL be applied, but using a maximum duration that ensures that the timestamp offset of the oldest generation of events in an RFC packet never exceeds 0x3FFF.

The RFC redundancy header timestamp offset value is only 14 bits, compared with the 16 bits in the event payload duration field. Since with other payloads the RTP timestamp typically increments for each new sample, the timestamp offset value becomes limiting on reported event duration. The limit becomes more constraining when older generations of events are also included in the combined payload. Retransmission of Final Packet The final packet for each event and for each segment SHOULD be sent a total of three times at the interval used by the source for updates.

This ensures that the duration of the event or segment can be recognized correctly even if an instance of the last packet is lost. A sender MAY use RFC [ 2 ] with up to two levels of redundancy to combine retransmissions with reports of new events, thus saving on header overheads. There is little point in sending initial or interim event reports redundantly because each succeeding packet describes the event fully except for typically irrelevant variations in volume.

A sender MAY delay setting the E bit until retransmitting the last packet for a tone, rather than setting the bit on its first transmission.

This avoids having to wait to detect whether the tone has indeed ended. Once the sender has set the E bit for a packet, it MUST continue to set the E bit for any further retransmissions of that packet. Packing Multiple Events into One Packet Multiple named events can be packed into a single RTP packet if and only if the events are consecutive and contiguous, i.

This approach is similar to having multiple frames of frame-based audio in one RTP packet. The constraint that packed events not overlap implies that events designated as states can be followed in a packet only by other state events that are mutually exclusive to them. The constraint itself is needed so that the beginning time of each event can be calculated at the receiver.

In a packet containing events packed in this way, the RTP timestamp MUST identify the beginning of the first event or segment in the packet. This will be true except when the packet carries the end of one segment and the beginning of the next segment of the same long-lasting event.

The E bit and duration for each event in the packet MUST be set using the same rules as if that event were the only event contained in the packet. Incrementing applies to retransmitted as well as initial instances of event reports, to permit the receiver to detect lost packets for RTP Control Protocol RTCP receiver reports. Receiving Procedures 2. SDP descriptions using the event payload MUST contain an fmtp format attribute that lists the event values that the receiver can process.

Since, for example, DTMF digit recognition takes several tens of milliseconds, the first few milliseconds of a digit will arrive as regular audio packets. Thus, careful time and power volume alignment between the audio samples and the events is needed to avoid generating spurious digits at the receiver. The receiver may also choose to delay playout of the tones by some small interval after playout of the preceding audio has ended, to ensure that downstream equipment can discriminate the tones properly.

Some implementations send events and encoded audio packets e. However, it is anticipated that these extra tones in general should not interfere with recognition at the far end. Receiver implementations MAY use different algorithms to create tones, including the two described here. Note that not all implementations have the need to re-create a tone; some may only care about recognizing the events. With either algorithm, a receiver may impose a playout delay to provide robustness against packet loss or delay.

The tradeoff between playout delay and other factors is discussed further in Section 2. As additional packets are received that extend the same tone, the waveform in the playout buffer is extended accordingly. Care has to be taken if audio is mixed, i. Thus, if a packet in a tone lasting longer than the packet interarrival time gets lost and the playout delay is short, a gap in the tone may occur.

Alternatively, the receiver can start a tone and play it until one of the following occurs: o it receives a packet with the E bit set; o it receives the next tone, distinguished by a different timestamp value noting that new segments of long-duration events also appear with a new timestamp value ; o it receives an alternative non-event media stream assuming none was being received while the event stream was active ; or o a given time period elapses.

This is more robust against packet loss, but may extend the tone beyond its original duration if all retransmissions of the last packet in an event are lost. Limiting the time period of extending the tone is necessary to avoid that a tone "gets stuck". This algorithm is not a license for senders to set the duration field to zero; it MUST be set to the current duration as described, since this is needed to create accurate events if the first event packet is lost, among other reasons.

A slight extension of tone durations and shortening of pauses is generally harmless. It MAY do so if the tone is of a type meant for human consumption or is one for which interruptions will not cause confusion at the receiving device. If a receiver receives an event packet for an event that it is not currently playing out and the packet does not have the M bit set, earlier packets for that event have evidently been lost. This can be confirmed by gaps in the RTP sequence number. In that case, further reports for the event MUST be ignored, as indicated in the previous paragraph.

If, on the other hand, the event has not been played out at all, the receiver MAY attempt to play the event out to the complete duration indicated in the event report. The appropriate behavior will depend on the event type, and requires consideration of the relationship of the event to audio media flows and whether correct event duration is essential to the correct operation of the media session.

Note: The jitter value should primarily be used as a means for comparing the reception quality between two users or two time periods, not as an absolute measure. If a zero volume is indicated for an event for which the volume field is defined, then the receiver MAY reconstruct the volume from the volume of non-event audio or MAY use the nominal value specified by the ITU Recommendation or other document defining the tone. Long-Duration Events If an event report is received with duration equal to the maximum duration expressible in the duration field 0xFFFF and the E bit for the report is not set, the event report may mark the end of a segment generated according to the procedures of Section 2.

The receiver uses the absence of a gap between the events to detect that it is receiving a single long-duration event. The total duration of a long-duration event is obviously the sum of the durations of the segments used to report it. This is equal to the duration of the final segment as indicated in the final packet for that segment , plus 0xFFFF multiplied by the number of segments preceding the final segment.

If a receiver is using the events component of the payload, event duration may be only an approximate indicator of division into segments, but the lack of an E bit and the adjacency of two reports with the same event code are strong indicators in themselves. Multiple Events in a Packet The procedures of Section 2. As a result, it is not strictly necessary for the receiver to know the start times of the events following the first one in order to play them out -- it needs only to respect the duration reported for each event.

Nevertheless, if knowledge of the start time for a given event after the first one is required, it is equal to the sum of the start time of the preceding event plus the duration of the preceding event. Soft States If the duration of a soft state event expires, the receiver SHOULD consider the value of the state to be "unknown" unless otherwise indicated in the event documentation. Congestion and Performance Packet transmission through the Internet is marked by occasional periods of congestion lasting on the order of second, during which network delay, jitter, and packet loss are all much higher than they are in between these periods.

Reference [28] characterizes this phenomenon. Well-behaved applications are expected, preferably, to reduce their demands on the network during such periods of congestion. At the least, they should not increase their demands. This section explores both application performance and the possibilities for good behavior in the face of congestion. Performance Requirements Typically, an implementation of the telephone-event payload will aim to limit the rate at which each of the following impairments occurs: a.

The relative importance of these constraints varies between applications. Reliability Mechanisms To improve reliability, all payload types including telephone-events can use a jitter buffer, i. This mechanism addresses the first four requirements listed above, but at the expense of the last one.

The named event procedures provide two complementary redundancy mechanisms to deal with lost packets: a. Intra-event updates: Events that last longer than one packetization period e. During an event, the RTP event payload format provides incremental updates on the event. The error resiliency afforded by this mechanism depends on whether the first or second algorithm in Section 2.

Repeat last event packet: As described in Section 2. This mechanism adds robustness to the reporting of the end of an event. It may be necessary to extend the level of redundancy to achieve requirement a in Section 2.

This is one more level of redundancy than required by the basic "Repeat last event packet" algorithm. Of course, the objective is probably unrealistically stringent; it was chosen to make a point. Where Section 2. This is done by using more than two levels of redundancy when necessary.

The use of RFC helps to mitigate the extra bandwidth demands that would be imposed simply by retransmitting final event packets more than three times. These two redundancy mechanisms clearly address requirement a in the previous section. They also help meet requirement c , to the extent that the redundant packets arrive before playout of the events they report is due to expire. They are not helpful in meeting the other requirements, although they do not directly cause impairments themselves in the way that a large jitter buffer increases end-to-end delay.

The playout algorithm is an additional mechanism for meeting the performance requirements. In particular, using the second algorithm in Section 2. Finally, there is an interaction between the packetization period used by a sender, the playout delay used by the receiver, and the vulnerability of an event flow to packet losses.

This improves end-to-end delays in applications where that matters. In view of the tradeoffs between the different reliability mechanisms, documentation of specific events SHOULD include a discussion of the appropriate design decisions for the applications of those events.

This mandate is repeated in the section on IANA considerations. Adjusting to Congestion So far, the discussion has been about meeting performance requirements. However, there is also the question of whether applications of events can adapt to congestion to the point that they reduce their demands on the networks during congestion.

In theory this can be done for events by increasing the packetization interval, so that fewer packets are sent per second. This has to be accompanied by an increased playout delay at the receiving end. Coordination between the two ends for this purpose is an interesting issue in itself. If it is done, however, such an action implies a one-time gap or extended playout of an event when the packetization interval is first extended, as well as increased end-to-end delay during the whole period of increased playout delay.

The benefit from such a measure varies primarily depending on the average duration of the events being handled. In the worst case, as a first example shows, the reduction in aggregate bandwidth usage due to an increased packetization interval may be quite modest. Suppose the average event duration is 3. Suppose further that four transmissions in total are required for a given event report to meet the loss objective.

Table 1 shows the impact of varying packetization intervals on the aggregate bit rate of the media stream. Extending the playout of a specific V. The reduction in number of packets per second with longer packetization periods is countered by the increase in packet size due to the increase in number of events per packet. For events of longer duration, the reduction in bandwidth is more proportional to the increase in packetization interval. The loss of final event reports may also be less critical, so that lower redundancy levels are acceptable.

Table 2 shows similar data to Table 1, but assuming ms events separated by 50 ms of silence as in an idealized DTMF-based text messaging session with only the basic two retransmissions for event completions.

No more than one level of redundancy is needed up to a packetization interval of 50 ms, although at that point most packets are reporting two events. Longer intervals require a second level of redundancy in at least some packets. DTMF digits may be consumed by entities such as gateways or application servers in the IP network, or by entities such as telephone switches or IVRs in the circuit switched network. The gateway likely has the necessary digital signal processors and algorithms, as it often needs to detect DTMF, e.

Having the gateway detect tones relieves the receiving Internet end system from having to do this work and also avoids having low bit-rate codecs like G. An Internet end system such as an "Internet phone" can emulate DTMF functionality without concerning itself with generating precise tone pairs and without imposing the burden of tone recognition on the receiver.

A similar distinction occurs at the receiving end. In the end system scenario, the DTMF events are consumed by the receiving entity itself. In the most common application, DTMF tones are sent in one direction only, typically from the calling end. The consuming device is most commonly an IVR.

DTMF may alternate with voice from either end. In most cases, the only constraint on tone duration is that it exceed a minimum value. Like this: Like Loading Short and informative article, thanks Andrew. I do these in my spare time and I am happy to hear that they are making a difference. You are correct and I fixed the text. Thanks for noticing it and commenting.

Hi Andrew, When we are dialing the verizon conferencing bridge number and when dialing the passcode it is detecting two tones for a single tone. Thank you for the explination. Very well written. I could help me what would be the difference? Am I correct in my thinking? I appreciate any help! Best Mihajlo. What would the represent? Someone should do a SIP trace to see if the digits are coming through. Hi Andrew, Firstly an excellent read — very clear and informative! During call set up I can see both offerings in their SDP profile, see below.

Look forward to our reply. Regards, David. Thanks for the prompt reply Andrew. Not use it. Send DTMF in the voice stream. Best Regards Fredrik. Hi Andrew, Thank you for explaining this. Thanks great article, I was Reading couple of article to get different between RFC out of band and in-band, but really I am not clear yet.

If someone can help me to be more clear will be great. Regards, Guillermo. Regards, GUillermo. Leave a Reply Cancel reply Enter your comment here Fill in your details below or click an icon to log in:. Email required Address never made public.

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