One document matched: draft-ietf-soc-overload-control-03.xml
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<front>
<title abbrev='Overload Control'>Session Initiation Protocol (SIP)
Overload Control</title>
<author initials="V.K." surname="Gurbani" fullname="Vijay K. Gurbani" role="editor">
<organization>Bell Laboratories, Alcatel-Lucent</organization>
<address>
<postal>
<street>1960 Lucent Lane, Rm 9C-533</street>
<city>Naperville</city> <region>IL</region>
<code>60563</code>
<country>USA</country>
</postal>
<email>vkg@bell-labs.com</email>
</address>
</author>
<author initials='V.H.' surname='Hilt' fullname='Volker Hilt'>
<organization>Bell Labs/Alcatel-Lucent</organization>
<address>
<postal>
<street>791 Holmdel-Keyport Rd</street>
<city>Holmdel</city> <region>NJ</region>
<code>07733</code>
<country>USA</country>
</postal>
<email>volkerh@bell-labs.com</email>
</address>
</author>
<author initials='H.S.' surname='Schulzrinne' fullname='Henning Schulzrinne'>
<organization abbrev='Columbia University'>Columbia University/Department
of Computer Science</organization>
<address>
<postal>
<street>450 Computer Science Building</street>
<city>New York</city> <region>NY</region>
<code>10027</code>
<country>USA</country>
</postal>
<phone>+1 212 939 7004</phone>
<email>hgs@cs.columbia.edu</email>
<uri>http://www.cs.columbia.edu</uri>
</address>
</author>
<date year='2011' />
<area>RAI</area>
<workgroup>SOC Working Group</workgroup>
<keyword>SIP</keyword>
<keyword>Overload Control</keyword>
<abstract>
<t>Overload occurs in Session Initiation Protocol (SIP) networks when
SIP servers have insufficient resources to handle all SIP messages
they receive. Even though the SIP protocol provides a limited
overload control mechanism through its 503 (Service Unavailable)
response code, SIP servers are still vulnerable to overload. This
document defines the behaviour of SIP servers involved in overload
control, and in addition, it specifies a loss-based overload scheme
for SIP.</t>
</abstract>
</front>
<middle>
<section title="Introduction">
<t>As with any network element, a Session Initiation Protocol
(SIP) <xref target="RFC3261" /> server can suffer from overload
when the number of SIP messages it receives exceeds the number of
messages it can process. Overload can pose a serious problem for a
network of SIP servers. During periods of overload, the throughput
of a network of SIP servers can be significantly degraded. In
fact, overload may lead to a situation in which the throughput
drops down to a small fraction of the original processing
capacity. This is often called congestion collapse.</t>
<t>Overload is said to occur if a SIP server does not have
sufficient resources to process all incoming SIP messages. These
resources may include CPU processing capacity, memory,
network bandwidth, input/output, or disk resources.</t>
<t>For overload control, we only consider failure cases where SIP
servers are unable to process all SIP requests due to resource
constraints. There are other cases where a SIP server can
successfully process incoming requests but has to reject them due
to failure conditions unrelated to the SIP server being overloaded.
For example, a PSTN gateway that runs out of trunks but
still has plenty of capacity to process SIP messages should
reject incoming INVITEs using a 488 (Not
Acceptable Here) response <xref target="RFC4412" />. Similarly, a
SIP registrar that has lost connectivity to its registration
database but is still capable of processing SIP requests should
reject REGISTER requests with a 500 (Server Error) response <xref
target="RFC3261" />. Overload control does not apply to these
cases and SIP provides appropriate response codes for them.</t>
<t>The SIP protocol provides a limited mechanism for overload
control through its 503 (Service Unavailable) response
code. However, this mechanism cannot prevent overload of a SIP
server and it cannot prevent congestion collapse. In fact, the use
of the 503 (Service Unavailable) response code may cause traffic
to oscillate and to shift between SIP servers and thereby worsen
an overload condition. A detailed discussion of the SIP overload
problem, the problems with the 503 (Service Unavailable) response
code and the requirements for a SIP overload control mechanism can
be found in <xref target="RFC5390" />.</t>
<t>This document defines the general behaviour of SIP servers and
clients involved in overload control in
<xref target="sec:gen-behaviour"/>. In addition,
<xref target="sec:loss-based"/> specifies a loss-based overload
control scheme. SIP clients and servers conformant to this
specification MUST implement the loss-based overload control
scheme. They MAY implement other overload control schemes as
well.</t>
</section>
<section title="Terminology">
<t>The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL
NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and
"OPTIONAL" in this document are to be interpreted as described in
<xref target="RFC2119">RFC 2119</xref>.</t>
<t>The normative statements in this specification as they apply
to SIP clients and SIP servers assume that both the SIP clients and
SIP servers support this specification. If, for instance, only a
SIP client supports this specification and not the SIP server,
then follows that the normative statements in this specification
pertinent to the behavior of a SIP server do not apply to the
server that does not support this specification.</t>
</section>
<section title="Overview of operations" anchor="sec:overview">
<t>We now explain the overview of how the overload control
mechanism operates by introducing the overload control parameters.
<xref target="sec:header"/> provides more details and
normative behavior on the parameters listed below.</t>
<t>Because overload control is best performed hop-by-hop, the
Via parameter is attractive since it allows two adjacent SIP
entities to indicate support for, and exchange information associated
with overload control. Additional advantages of this choice are
discussed in <xref target="sec:dc:sm:response"/>. An alternative
mechanism using SIP event packages was also considered, and the
characteristics of that choice are further outlined in <xref target=
"sec:dc:sm:event"/>.</t>
<t>This document defines four new parameters for the SIP Via header
for overload control. These parameters provide a mechanism for
conveying overload control information between adjacent SIP
entities. The "oc" parameter is used by a SIP server to indicate
a reduction in the amount of requests arrving at the server. The
"oc-algo" parameter contains a token or a list of tokens corresponding
to the class of overload control algorithms supported by the client.
The server chooses one algorithm from this list. The "oc-validity"
parameter establishes a time limit for which overload control is
in effect, and the "oc-seq" parameter aids in sequencing the
responses at the client. These parameters are discussed in detail
in the next section.</t>
</section> <!-- sec:overview -->
<section title="Via Header Parameters for Overload Control"
anchor="sec:header">
<t>The four Via header parameters are introduced below.
Further context in how to interpret these under various
conditions is provided in <xref target="sec:gen-behaviour"/>.</t>
<section title="The oc paramater" anchor="oc-param">
<t>This parameter is inserted by the SIP client and updated by the
SIP server.</t>
<t>A SIP client MUST add
an "oc" parameter to the topmost Via header it inserts into the
SIP request. This provides an indication to downstream neighbors
that this server supports overload control. When inserted into
a request by a SIP client to indicate support for overload
control, there MUST NOT be a value associated with the
parameter.</t>
<t>The downstream server MUST add a value to the "oc" parameter
in the response going upstream; this value represents a metric
by which the requests arriving at the downstream server should
be reduced.</t>
<t>When a SIP client receives a response with the value in
the "oc" parameter filled in, it SHOULD reduce, by the amount
indicated, the number of requests going downstream to the
SIP server from which it received the response (see
<xref target="sec:respond-overload"/> for pertinent discussion
on traffic reduction).</t>
</section> <!-- oc-param -->
<section title="The oc-algo parameter" anchor="oc-algo">
<t>This parameter is inserted by the SIP client and updated by the
SIP server.</t>
<t>A SIP client MAY add an "oc-algo" parameter to the topmost
Via header it inserts into the SIP request. This parameter
contains a comma-separated list of overload control algorithms.
A SIP client conformant to this specification MUST support the
loss-based overload control scheme and MUST insert the token
"loss" as the "oc-algo" parameter value. In addition, the
SIP client MAY insert other tokens in the "oc-algo" parameter valus
if it supports other overload control schemes, such as a rate-based
scheme (<xref target="I-D.soc-rate"/>). Each element in
the comma-separated
list corresponds to the class of overload control algorithms
supported by the SIP client. When a downstream SIP server
receives a request with a choice of overload control algorithms
specified in the "oc-algo" parameter value, it MUST choose one
algorithm from the list and MUST pare the list down to include
the one chosen algorithm. The pared down list consisting of the
chosen algorithm MUST be returned to the upstream SIP client in the
response.</t>
<t>Once a SIP client and a SIP server have converged to a
mutually agreeable class of overload control algorithm, the
agreed upon class stays in effect for a non-trivial duration
of time to allow the overload control algorithm to stabilize
its behaviour (see <xref target="sec:stabilize"/>).</t>
</section> <!-- oc-algo -->
<section title="The oc-validity parameter" anchor="oc-validity">
<t>This parameter is inserted by the SIP server.</t>
<t>This parameter contains a value that indicates an interval of time
(measured in milliseconds) that the load reduction specified
value of the "oc" parameter should be in effect. The default
value of the "oc-validity" parameter is 500 (millisecond).</t>
<t>A value of 0 in the "oc-validity" parameter is reserved
to denote the event that the server wishes to stop overload
control (see <xref target="sec:terminate"/> for more information).</t>
<t>A non-zero value for the "oc-validity" parameter MUST only be
present in conjunction with an "oc" parameter.</t>
</section> <!-- oc-validity -->
<section title="The oc-seq parameter" anchor="oc-seq">
<t>This parameter is inserted by the SIP server.</t>
<t>This parameter contains a value that indicates the sequence
number associated with the "oc" parameter. Some implementations
may be capable of updating the overload control information before
the validity period specified by the "oc-validity" parameter
expires. Such implementations MUST have an increasing value in
the "oc-seq" parameter for each response sent to the upstream
SIP client. This is to allow the upstream SIP client to properly
collate out-of-order responses.</t>
</section> <!-- oc-seq -->
<!--
<t><list>
<t>OPEN ISSUE: To throttle upstream neighbors in a fair way,
it is important that a SIP server can estimate the load each
upstream neighbor receives for this server before it is
throttled. This enables the server to throttle each upstream
neighbor in the same way and thus provides each request the
same chance of succeeding. In rate- and window-based overload
control systems, a SIP server does not know how many messages
each upstream neighbor had received for the server before
throttling took place. A solution to this problem is to allow
servers to report the unthrottled load for a downstream neighbor
in the 'oc_accept' parameter.</t>
</list></t>
-->
</section> <!-- sec:header -->
<section title="General behaviour" anchor="sec:gen-behaviour">
<section title="Creating and updating the overload control parameters"
anchor="oc-create">
<t>A SIP server can provide overload control feedback to its
upstream neighbors by providing a value for the "oc" parameter
to the topmost Via header field of a SIP response. The topmost
Via header is determined after the SIP server has removed its
own Via header; i.e., it is the Via header that was generated
by the upstream neighbor.</t>
<t>Since the topmost Via header of a response will be removed by
an upstream neighbor after processing it, overload control
feedback contained in the "oc" parameter will not travel beyond
the upstream SIP client. A Via header parameter therefore provides
hop-by-hop semantics for overload control feedback (see <xref
target="I-D.ietf-soc-overload-design" />) even if the
next hop neighbor does not support this specification.</t>
<t>The "oc" parameter can be used in all response types,
including provisional, success and failure responses (please see
<xref target="sec:100"/> for special consideration on
transporting overload control parameters in a 100-Trying response). A SIP
server MAY update the "oc" parameter in all responses it
is sending. A SIP server MUST update the "oc" parameter to responses
when the transmission of overload control feedback is required
by the overload control algorithm to limit the traffic received
by the server. I.e., a SIP server MUST update the "oc" parameter
when the overload control algorithm sets the value of an "oc"
parameter to a value different than the default value.</t>
<t>A SIP server that has updated the "oc" parameter to Via header
SHOULD also add a "oc-validity" parameter to the same Via
header. The "oc-validity" parameter defines the time in
milliseconds during which the content (i.e., the overload
control feedback) of the "oc" parameter is valid. The default
value of the "oc-validity" parameter is 500 (millisecond). A SIP
server SHOULD use a shorter "oc-validity" time if its overload
status varies quickly and MAY use a longer "oc-validity" time
if this status is more stable. If the "oc-validity" parameter is
not present, its default value is used. The "oc-validity"
parameter MUST NOT be used in a Via header that did not
originally contain an "oc" parameter when received. Furthermore,
when a SIP server receives a request with the topmost Via
header containing only an "oc-validity" parameter without
the accompanying "oc" parameter, it MUST ignore the "oc-validity"</t>
<t>When a SIP server retransmits a response, it SHOULD
use the "oc" parameter value and "oc-validity" parameter
value consistent with the overload state at the time the
retransmitted response is sent. This implies that the
values in the "oc" and "oc-validity" parameters may be
different then the ones used in previous retransmissions
of the response. Due to the fact that responses sent over
UDP may be subject to delays in the network and arrive
out of order, the "oc-seq" parameter aids in detecting a
stale "oc" parameter value.</t>
<t>Implementations that are capable of updating the "oc"
and "oc-validity" parameter values for retransmissions MUST
insert the "oc-seq" parameter. The value of this parameter
MUST be a set of numbers drawn from an increasing sequence.</t>
<t>Implementations that are not capable of updating the "oc"
and "oc-validity" parameter values for retransmissions --- or
implementations that do not want to do so because they will
have to regenerate the message to be retransmitted --- MUST
still insert a "oc-seq" parameter in the first response
associated with a transaction; however, they do not have to
update the value in subsequent retransmissions.</t>
<!-- Ed. note: The discussion that lead to the above is
captured in
http://www.ietf.org/mail-archive/web/sip-overload/
current/msg00250.html
-->
<t>The "oc-validity" and "oc-seq" Via header
parameters are only defined in SIP responses and MUST NOT be
used in SIP requests. These parameters are only useful to the
upstream neighbor of a SIP server (i.e., the entity that is
sending requests to the SIP server) since this is the entity
that can offload traffic by redirecting/rejecting new requests.
If requests are forwarded in both directions between two SIP
servers (i.e., the roles of upstream/downstream neighbors
change), there are also responses flowing in both
directions. Thus, both SIP servers can exchange overload
information.</t>
<!--
Not sure the paragraph below adds anything useful to the
discussion. I will comment it out right now and see if
someone notices its disappearance. vkg Nov-19-2010.
<t>While adding "oc" and "oc-validity" parameters to
requests may increase the frequency with which overload
information is exchanged in these scenarios, this increase will
rarely provide benefits and does not justify the added overhead
and complexity needed.</t>
-->
<t>Since overload control protects a SIP server from overload,
it is RECOMMENDED that a SIP server use the mechanisms described
in this specification. However, if a SIP server wanted to limit its
overload control capability for privacy reasons, it MAY decide to
perform overload control only for requests that are received
on a secure transport channel, such as TLS. This enables a SIP
server to protect overload control information and ensure that
it is only visible to trusted parties. </t>
</section>
<section title="Determining the 'oc' Parameter Value"
anchor="oc-determine">
<t>The value of the "oc" parameter is determined by the
overloaded server using any pertinent information at its
disposal. The process by which an overloaded server
determines the value of the "oc" parameter is considered
out of scope for this document.</t>
</section>
<section title="Processing the Overload Control Parameters"
anchor="oc-process">
<t>A SIP client compliant to this specification SHOULD remove
"oc", "oc-validity" and "oc-seq" parameters from all Via headers of a
response received, except for the topmost Via header. This
prevents overload control parameters that were accidentally or
maliciously inserted into Via headers by a downstream SIP server
from traveling upstream.</t>
<t>A SIP client maintains the "oc" parameter values received
along with the address and port number of the SIP servers from
which they were received for the duration specified in the
"oc-validity" parameter or the default duration. Each time a
SIP client receives a response with an "oc" parameter from a
downstream SIP server, it overwrites the "oc" value it has
currently stored for this server with the new value received.
The SIP client restarts the validity period of an "oc" parameter
each time a response with an "oc" parameter is received from
this server. A stored "oc" parameter value MUST be discarded once
it has reached the end of its validity.</t>
</section>
<section title="Using the Overload Control Parameter Values"
anchor="oc-use">
<t>A SIP client compliant to this specification MUST honor
overload control values it receives from downstream neighbors. The
SIP client MUST NOT forward more requests to a SIP server than
allowed by the current "oc" parameter value from a particular
downstream server.</t>
<t>When forwarding a SIP request, a SIP client uses the SIP
procedures of <xref target="RFC3263"/> to determine the next
hop SIP server. The procedures of <xref target="RFC3263"/> take
as input a SIP URI, extract the
domain portion of that URI for use as a lookup key, and query
the Domain Name Service (DNS) to obtain an ordered set of one
or more IP addresses with a port number and transport corresponding
to each IP address in this set (the "Expected Output").</t>
<t>After selecting a specific SIP server from the Expected Output,
the SIP client MUST determine if it already has overload control
parameter values for the server chosen from the Expected Output.
If the SIP client has a non-expired "oc" parameter value for
the server chosen from the Expected Output, and this chosen server
is operating in overload control mode. Thus, the SIP client MUST
determine if it can or cannot forward the current request to the
SIP server depending on the nature of the request and the
prevailing overload conditions.</t>
<t>The particular algorithm used to determine whether or not to
forward a particular SIP request is a matter of local policy,
and may take into account a variety of prioritization factors.
However, this local policy SHOULD generate the same number of
SIP requests as the default algorithm defined by the overload
control scheme being used.</t>
<!--
<t>The SIP client SHOULD use the following algorithm to determine
if it can forward the request. The SIP client draws a random
number between 1 and 100 for the current request. If the random
number is less than or equal to the 'oc' parameter value, the
request is not forwarded. Otherwise, the request is
forwarded (note that this algorithm does not prioritize the
requests it is dropping --- c.f. OPEN ISSUE 3 in <xref target=
"msg-priority"/>.) Any other algorithms that
approximate the random number algorithm may be used as well.</t>
-->
<!--
<t><list>
<t>OPEN ISSUE: the specific mechanisms to throttle traffic
depend on the type of feedback conveyed in the 'oc' parameter
value. It needs to be defined depending on whether a
loss-based, rate-based or window-based feedback is used.</t>
</list></t>
<t>The treatment of SIP requests that cannot be forwarded to the
selected SIP Server is a matter of local policy. A SIP client
MAY try to find an alternative target or it MAY reject the
request (see <xref target="sec:5xx" />).</t>
-->
</section>
<section title="Forwarding the overload control parameters"
anchor="sec:forward">
<t>A SIP client MAY forward the content of an "oc" parameter it
has received from a downstream neighbor on to its upstream
neighbor. However, forwarding the content of the "oc" parameter
is generally NOT RECOMMENDED and should only be performed if
permitted by the configuration of SIP servers. For example, a
SIP server that only relays messages between exactly two SIP
servers may forward an "oc" parameter. The "oc" parameter is
forwarded by copying it from the Via in which it was received
into the next Via header (i.e., the Via header that will be on
top after processing the response). If an "oc-validity"
parameter is present, MUST be copied along with the "oc"
parameter.</t>
</section> <!-- sec:forward -->
<section title="Terminating overload control"
anchor="sec:terminate">
<t>A SIP client stops applying overload control to the number of
messages forwarded (i.e., it stops reducing the number of messages
forwarded) if one of the following events occur:</t>
<t>
<list style="numbers">
<t>The "oc" parameter is set to a value that allows the
client to forward all traffic;</t>
<t>The "oc-validity" period negotiated to put the server
and client in overload state expires;</t>
<t>The client is explicitly told by the server to
stop performing overload control using the "oc-validity=0"
parameter.</t>
</list>
</t>
<t>A SIP server can decide to terminate overload control by explicitly
signaling the client. To do so, the SIP server MUST set the value
of the "oc-validity" parameter to 0. The SIP server MUST increment
the value of "oc-seq", and SHOULD set the value of the "oc" parameter
to 0.
<list>
<t></t>
<t>Note that the loss-based overload control scheme
(<xref target="sec:loss-based"/>) can
effectively stop overload control by setting the value of the
"oc" parameter to 0. However, the rate-based scheme
(<xref target="I-D.soc-rate"/>) needs an
additional piece of information in the form of "oc-validity=0".</t>
</list>
</t>
<t>When the client receives a response with a higher "oc-seq"
number than the one it currently is processing, it checks
the "oc-validity" parameter. If the value of the "oc-validity"
parameter is 0, the client MUST stop performing overload control of
messages destined to the server and the traffic should flow without
any reduction. Furthermore, when the value of the "oc-validity"
parameter is 0, the client SHOULD disregard the value in the
"oc" parameter.</t>
</section> <!-- sec:terminate -->
<section title="Stabilizing overload control"
anchor="sec:stabilize">
<t>Realities of deployments of SIP necessitate that the overload control
algorithm be renegotiated upon a system reboot or a software upgrade.
However, frequent renegotiations of the overload control algorithm
MUST be avoided. A rapid renegotiation of the overload
control algorithm will not benefit the client or the server as such
flapping does not allow the chosen algorithm to measure and fine tune
its behavior over a period of time. Renegotiation, when desired,
is simply accomplished by the SIP client sending a complete list
of overload control algorithms it supports in a "oc-algo" parameter
in a request going downstream. The downstream server, as before,
MUST choose one algorithm from the list and MUST pare
the list down to include the one chosen algorithm. The pared
down list consisting of the chosen algorithm MUST be returned
to the upstream SIP client in the response and stays in
effect until the next renegotiation.</t>
<t>Once the client and server agree on an overload control
algorithm, it MUST remain in effect for at least 3600 seconds
(1 hour) before renegotiation occurs.
<list>
<t></t>
<t>One way to accomplish this involves the client
saving the time of the last negotiation in a lookup
table, indexed by the server's network identifiers.
Renegotiation is only done when the time of the last
negotiation has surpassed 3600 seconds.</t>
</list>
</t>
</section> <!-- sec:stabilize -->
<section title="Self-Limiting" anchor="sec:self-limiting">
<t>In some cases, a SIP client may not receive a response from a
server after sending a request. <xref
target="RFC3261">RFC3261</xref> defines that when a timeout
error is received from the transaction layer, it MUST be treated
as if a 408 (Request Timeout) status code has been received. If
a fatal transport error is reported by the transport layer, it
MUST be treated as a 503 (Service Unavailable) status code.</t>
<t>In the event of repeated timeouts or fatal transport errors,
the SIP client MUST stop sending requests to this server. The
SIP client SHOULD occasionally forward a single request to probe
if the downstream server is alive. Once a SIP client has successfully
transmitted a request to the downstream server, the SIP client
can resume normal traffic rates. It should, of course, honor
any "oc" parameters it may receive subsequent to resuming normal
traffic rates.</t>
<!--
<t><list>
<t>OPEN ISSUE: If a downstream neighbor does not respond to
a request at all, the upstream SIP client will stop sending
requests to the downstream neighbor. The upstream SIP client
will periodically forward a single request to probe the
health of its downstream neighbor. It has been suggested ---
see
http://www.ietf.org/mail-archive/web/sip-overload/current/msg00229.html
--- that we have a notification mechanism in place for the
downstream neighbor to signal to the upstream SIP client that it
is ready to receive requests. This notification scheme has
advantages, but comes with obvious disadvantages as well. Need
some more discussion around this.</t>
</list></t>
<t><list>
<t>OPEN ISSUE: waiting for a timeout to occur seems a long time
before starting to throttle back. It could make sense to
throttle back earlier if no response is received for requests
transmitted.</t>
</list></t>
-->
</section>
<section title="Responding to an Overload Indication"
anchor="sec:respond-overload">
<t>A SIP client can receive overload control feedback indicating
that it needs to reduce the traffic it sends to its downstream
server. The client can accomplish this task by sending some of
the requests that would have gone to the overloaded element to a
different destination. It needs to ensure, however, that this
destination is not in overload and capable of processing the
extra load. A client can also buffer requests in the hope that
the overload condition will resolve quickly and the requests
still can be forwarded in time. In many cases, however, it will
need to reject these requests.</t>
<section title="Message prioritization at the hop before the
overloaded server" anchor="msg-priority">
<t>During an overload condition, a SIP client needs to
prioritize requests and select those requests that need to be rejected
or redirected. While this selection is largely a matter of local
policy, certain heuristics can be suggested. One, during overload
control, the SIP client should preserve existing dialogs as much
as possible. This suggests that mid-dialog requests MAY be
given preferential treatment. Similarly, requests that result in
releasing resources (such as a BYE) MAY also be given
preferential treatment.</t>
<t>A SIP client SHOULD honor the local policy for prioritizing
SIP requests such as policies based on the content of the
Resource-Priority header (RPH, <xref target="RFC4412">RFC4412</xref>).
Specific (namespace.value) RPH contents may indicate high priority
requests that should be preserved as much as possible during
overload. The RPH contents can also indicate a low-priority
request that is eligible to be dropped during times of overload.
Other indicators, such as the SOS URN <xref target="RFC5031"/>
indicating an emergency request, may also be used for
prioritization.</t>
<t>Local policy could also include giving precedence to
mid-dialog SIP requests (re-INVITEs, UPDATEs, BYEs etc.) in times
of overload. A local policy can be expected to combine both
the SIP request type and the prioritization markings, and SHOULD
be honored when overload conditions prevail.</t>
<t>A SIP client SHOULD honor user-level load control filters
installed by signaling neighbors
<xref target="I-D.ietf-soc-load-control-event-package"/> by sending
the SIP messages that matched the filter downstream.</t>
<!--
<t><list>
<t>OPEN ISSUE 3: The process by which the upstream server
selects messages to be rejected to reduce load needs to be
discussed further. Clearly, SIP messages that contain the
Resource-Priority header should not be rejected. Also, mid-
dialog requests (re-INVITEs, UPDATEs, BYEs etc.) should be
honored, if possible. This can be left as a policy decision
with guidelines provided --- example, if a request has both
the To tag and From tag, do not drop it since it is a mid-
dialog request; do not drop requests with the Resource-Priority
header, etc. Some discussion on this is captured in
http://www.ietf.org/mail-archive/web/sip-overload/current/msg00272.html.
</t>
<t></t>
<t>This issue also has a bearing on the default algorithm to be
outlined in Section <xref target="oc-use"/>.</t>
</list></t>
-->
</section>
<section title="Rejecting requests at an overloaded server"
anchor="sec:5xx">
<t>If the upstream SIP client to the overloaded server does
not support overload control, it will continue to direct
requests to the overloaded server. Thus, the overloaded
server must bear the cost of rejecting some session requests as
well as the cost of processing other requests to completion. It
would be fair to devote the same amount of processing at the
overloaded server to the combination of rejection and processing
as the overloaded server would devote to processing requests
from an upstream SIP client that supported overload control.
This is to ensure that SIP servers that do not support this
specification don't receive an unfair advantage over those that
do. </t>
<t>A SIP server that is under overload and has started to
throttle incoming traffic MUST reject this request with a
"503 (Service Unavailable)" response without Retry-After header
to reject a fraction of requests from upstream neighbors that do
not support overload control.</t>
</section>
</section>
<section title="100-Trying provisional response and overload control
parameters" anchor="sec:100">
<t>The overload control information sent from a SIP server to a client
is transported in the responses. While implementations can insert overload
control information in any response, special attention should be accorded
to overload control information transported in a 100-Trying response.</t>
<t>Traditionally, the 100-Trying response has been used in SIP to quench
retransmissions. In some implementations, the 100-Trying message may
not be generated by the transaction user (TU) nor consumed by the TU.
In these implementations, the 100-Trying response is generated at the
transaction layer and sent to the upstream SIP client. At the receiving
SIP client, the 100-Trying is consumed at the transaction layer by
inhibiting the retransmission of the corresponding request. Consequently,
implementations that insert overload control information in the 100-Trying
cannot assume that the upstream SIP client passed the overload control
information in the 100-Trying to their corresponding TU. For this reason,
implementations that insert overload control information in the 100-Trying
MUST re-insert the same (or updated) overload control information in the
first non-100 response being sent to the upstream SIP client.</t>
<!-- c.f. http://www.ietf.org/mail-archive/web/sip-overload/
current/msg00452.html -->
</section> <!-- sec:100 -->
</section> <!-- sec:gen-behaviour -->
<section title="The loss-based overload control scheme"
anchor="sec:loss-based">
<t>A loss percentage enables a SIP server to ask an upstream neighbor to
reduce the number of requests it would normally forward to this
server by X%. For example, a SIP server can ask an upstream neighbor
to reduce the number of requests this neighbor would normally send by
10%. The upstream neighbor then redirects or rejects 10% of the
traffic that is destined for this server.</t>
<t>This section specifies the semantics of the overload control
parameters associated with the loss-based overload control scheme.
The general behaviour of SIP clients and servers is specified in
<xref target="sec:gen-behaviour"/> and is applicable to SIP clients
and servers that implement loss-based overload control.</t>
<section title="Special parameter values for loss-based overload
control" anchor="sec:loss-based-params">
<t>The loss-based overload control scheme is identified using the
token "loss". This token MUST appear in the "oc-algo" parameter.</t>
<t>A SIP server, upon entering the overload state, will assign a
value to the "oc" parameter. This value MUST be restricted in
the range of [0, 100], inclusive. This value MUST be interpreted
as a percentage, and the SIP client MUST reduce the number of
requests being forwarded to the overloaded server by that amount.
The SIP client may use any algorithm that reduces the traffic
arriving at the overloaded server by the amount indicated. Such
an algorithm SHOULD honor the message prioritization discussion
of <xref target="msg-priority"/>. While a particular algorithm
is not subject to standardization, for completeness a default
algorithm for loss-based overload control is provided in
<xref target="sec:default-loss-algo"/>.</t>
<t>When a SIP server receives a request from a client with an
"oc" parameter but without a value, and the SIP server is
not experiencing overload, it MUST assign a value of 0 to the
"oc" parameter in the response. Assigning such a value lets
the client know that the server supports overload control and
is not currently experiencing overload.</t>
<t>When the "oc-validity" parameter is used to signify overload
control termination (<xref target="sec:terminate"/>), the server
MUST insert a value of 0 in the "oc-validity" parameter. The
server MUST insert a value of 0 in the "oc" parameter as well.
When a client receives a response whose "oc-validity" parameter
contains a 0, it MUST treat any non-zero value in the "oc"
parameter as if it had received a value of 0 in that parameter.</t>
</section> <!-- sec:loss-based-params -->
<section title="Example" anchor="sec:example">
<t>Consider a SIP client, P1, which is sending requests to another
downstream SIP server, P2. The following snippets of SIP messages
demonstrate how the overload control parameters work.</t>
<figure><artwork><![CDATA[
INVITE sips:user@example.com SIP/2.0
Via: SIP/2.0/TLS p1.example.net;
branch=z9hG4bK2d4790.1;received=192.0.2.111;oc;
oc-algo="loss,A"
...
SIP/2.0 100 Trying
Via: SIP/2.0/TLS p1.example.net;
branch=z9hG4bK2d4790.1;received=192.0.2.111;
oc=0;oc-algo="loss";
...
]]></artwork></figure>
<t>In the messages above, the first line is sent by P1 to P2.
This line is a SIP request; because P1 supports overload control,
it inserts the "oc" parameter in the topmost Via header that it
created. P1 supports two overload control algorithms: loss and
some algorithm called "A".</t>
<t>The second line --- a SIP response --- shows the topmost Via
header amended by P2 according to this specification and sent to
P1. Because P2 also supports overload control, it chooses the
"loss" based scheme and sends that back to P1 in the "oc-algo"
parameter. It also sets the value of "oc" parameter to 0.</t>
<t>Had P2 not supported overload control, it would have left
the "oc" and "oc-algo" parameters unchanged, thus allowing the
client to know that it did not support overload control.</t>
<t>At some later time, P2 starts to experience overload. It
sends the following SIP message indicating that P1 should
decrease the messages arriving to P2 by 20% for 1s.</t>
<figure><artwork><![CDATA[
SIP/2.0 180 Ringing
Via: SIP/2.0/TLS p1.example.net;
branch=z9hG4bK2d4790.3;received=192.0.2.111;
oc=20;oc-algo="loss";oc-validity=1000;
oc-seq=1282321615.782
...
]]></artwork></figure>
<t>After 500ms, the overload condition at P2 subsides. It then
sends out the message below to allow P1 to send all messages
destined to P2.</t>
<figure><artwork><![CDATA[
SIP/2.0 183 Queued
Via: SIP/2.0/TLS p1.example.net;
branch=z9hG4bK2d4790.4;received=192.0.2.111;
oc=0;oc-algo="loss";oc-validity=0;oc-seq=1282321887.783
...
]]></artwork></figure>
</section> <!-- sec:example -->
<section title="Default algorithm for loss-based overload control"
anchor="sec:default-loss-algo">
<t>To pull in the algorithm from discussion on working group
list (see www.ietf.org/mail-archive/web/sip-overload/current/msg00318.html).</t>
</section> <!-- sec:default-loss-based-algo -->
</section> <!-- sec:loss-based -->
<section title="Relationship with other IETF SIP load control efforts"
anchor="sec:relationship">
<t>The overload control mechanism described in this document is reactive
in nature and apart from message prioritization directives listed in
<xref target="msg-priority"/> the mechanisms described in this draft
will not discriminate requests based on user identity, filtering action
and arrival time. SIP networks that require pro-active overload control
mechanisms can upload user-level load control filters as described in
<xref target="I-D.ietf-soc-load-control-event-package"/>.</t>
</section> <!-- sec:relationship -->
<section title="Syntax" anchor="sec:syntax">
<!--
<t><list>
<t>OPEN ISSUE: the syntax of the 'oc' Via header parameter
depends on the overload control method (i.e., loss-based,
rate-based or window-based) in use. The following syntax
definition defines a rate-based 'oc' header. This syntax needs
to be adjusted if rate-based or window-based overload control is
used.</t>
</list></t>
-->
<t>This specification extends the existing definition of the Via
header field parameters of <xref target="RFC3261"/> as follows:</t>
<figure>
<artwork><![CDATA[
via-params = via-ttl / via-maddr
/ via-received / via-branch
/ oc / oc-validity
/ oc-seq / oc-algo / via-extension
oc = "oc" [EQUAL oc-num]
oc-num = 1*DIGIT
oc-validity = "oc-validity" [EQUAL delta-ms]
oc-seq = "oc-seq" EQUAL 1*12DIGIT "." 1*5DIGIT
oc-algo = "oc-algo" EQUAL DQUOTE algo-list *(COMMA algo-list)
DQUOTE
algo-list = "loss" / *(other-algo)
other-algo = %x41-5A / %x61-7A / %x30-39
]]></artwork>
</figure>
</section>
<section title="Design Considerations" anchor="sec:dc">
<t>This section discusses specific design considerations for the
mechanism described in this document. General design
considerations for SIP overload control can be found in
<xref target="I-D.ietf-soc-overload-design" />.</t>
<section title="SIP Mechanism" anchor="sec:dc:sm">
<t>A SIP mechanism is needed to convey overload feedback from
the receiving to the sending SIP entity. A number of different
alternatives exist to implement such a mechanism.</t>
<section title="SIP Response Header" anchor="sec:dc:sm:response">
<t>Overload control information can be transmitted using a new
Via header field parameter for overload control. A SIP server
can add this header parameter to the responses it is sending
upstream to provide overload control feedback to its upstream
neighbors. This approach has the following
characteristics:</t>
<t><list style='symbols'>
<t>A Via header parameter is light-weight and creates very
little overhead. It does not require the transmission of
additional messages for overload control and does not
increase traffic or processing burdens in an overload
situation. </t>
<t>Overload control status can frequently be reported to
upstream neighbors since it is a part of a SIP
response. This enables the use of this mechanism in
scenarios where the overload status needs to be adjusted
frequently. It also enables the use of overload control
mechanisms that use regular feedback such as window-based
overload control.</t>
<t>With a Via header parameter, overload control status is
inherent in SIP signaling and is automatically conveyed to
all relevant upstream neighbors, i.e., neighbors that are
currently contributing traffic. There is no need for a SIP
server to specifically track and manage the set of current
upstream or downstream neighbors with which it should
exchange overload feedback.</t>
<t>Overload status is not conveyed to inactive senders. This
avoids the transmission of overload feedback to inactive
senders, which do not contribute traffic. If an inactive
sender starts to transmit while the receiver is in overload
it will receive overload feedback in the first response and
can adjust the amount of traffic forwarded accordingly. </t>
<t>A SIP server can limit the distribution of overload
control information by only inserting it into responses to
known upstream neighbors. A SIP server can use transport
level authentication (e.g., via TLS) with its upstream
neighbors.</t>
</list></t>
</section>
<section title="SIP Event Package" anchor="sec:dc:sm:event">
<t>Overload control information can also be conveyed from a
receiver to a sender using a new event package. Such an event
package enables a sending entity to subscribe to the overload
status of its downstream neighbors and receive notifications
of overload control status changes in NOTIFY requests. This
approach has the following characteristics:</t>
<t><list style='symbols'>
<t>Overload control information is conveyed decoupled from
SIP signaling. It enables an overload control manager, which
is a separate entity, to monitor the load on other servers
and provide overload control feedback to all SIP servers
that have set up subscriptions with the controller.</t>
<t>With an event package, a receiver can send updates to
senders that are currently inactive. Inactive senders will
receive a notification about the overload and can refrain
from sending traffic to this neighbor until the overload
condition is resolved. The receiver can also notify all
potential senders once they are permitted to send traffic
again. However, these notifications do generate additional
traffic, which adds to the overall load.</t>
<t>A SIP entity needs to set up and maintain overload
control subscriptions with all upstream and downstream
neighbors. A new subscription needs to be set up
before/while a request is transmitted to a new downstream
neighbor. Servers can be configured to subscribe at boot
time. However, this would require additional protection to
avoid the avalanche restart problem for overload
control. Subscriptions need to be terminated when they are
not needed any more, which can be done, for example, using a
timeout mechanism.</t>
<t>A receiver needs to send NOTIFY messages to all
subscribed upstream neighbors in a timely manner when the
control algorithm requires a change in the control variable
(e.g., when a SIP server is in an overload condition). This
includes active as well as inactive neighbors. These NOTIFYs
add to the amount of traffic that needs to be processed. To
ensure that these requests will not be dropped due to
overload, a priority mechanism needs to be implemented in
all servers these request will pass through.</t>
<t>As overload feedback is sent to all senders in separate
messages, this mechanism is not suitable when frequent
overload control feedback is needed.</t>
<t>A SIP server can limit the set of senders that can
receive overload control information by authenticating
subscriptions to this event package.</t>
<t>This approach requires each proxy to implement user agent
functionality (UAS and UAC) to manage the subscriptions.</t>
</list></t>
</section>
</section>
<section title="Backwards Compatibility">
<t>An new overload control mechanism needs to be backwards
compatible so that it can be gradually introduced into a network
and functions properly if only a fraction of the servers support
it.</t>
<t>Hop-by-hop overload control (see
<xref target="I-D.ietf-soc-overload-design" />) has the
advantage that it does not require that all SIP entities in a
network support it. It can be used effectively between two
adjacent SIP servers if both servers support overload control
and does not depend on the support from any other server or user
agent. The more SIP servers in a network support hop-by-hop
overload control, the better protected the network is against
occurrences of overload.</t>
<t>A SIP server may have multiple upstream neighbors from which
only some may support overload control. If a server would simply
use this overload control mechanism, only those that support it
would reduce traffic. Others would keep sending at the full rate
and benefit from the throttling by the servers that support
overload control. In other words, upstream neighbors that do not
support overload control would be better off than those that
do.</t>
<t>A SIP server should therefore use 5xx responses towards
upstream neighbors that do not support overload control. The
server should reject the same amount of requests with 5xx
responses that would be otherwise be rejected/redirected by the
upstream neighbor if it would support overload control. If the
load condition on the server does not permit the creation of 5xx
responses, the server should drop all requests from servers that
do not support overload control.</t>
</section>
</section>
<section anchor="sec:security" title="Security Considerations">
<t>Overload control mechanisms can be used by an attacker to
conduct a denial-of-service attack on a SIP entity if the attacker
can pretend that the SIP entity is overloaded. When such a forged
overload indication is received by an upstream SIP client, it will
stop sending traffic to the victim. Thus, the victim is
subject to a denial-of-service attack.</t>
<t>An attacker can create forged overload feedback by inserting
itself into the communication between the victim and its
upstream neighbors. The attacker would need to add overload
feedback indicating a high load to the responses passed from the
victim to its upstream neighbor. Proxies can prevent this attack by
communicating via TLS. Since overload feedback has no meaning
beyond the next hop, there is no need to secure the communication
over multiple hops.</t>
<t>Another way to conduct an attack is to send a message
containing a high overload feedback value through a proxy that does not
support this extension. If this feedback is added to the second Via
headers (or all Via headers), it will reach the next upstream
proxy. If the attacker can make the recipient believe that the
overload status was created by its direct downstream neighbor (and
not by the attacker further downstream) the recipient stops
sending traffic to the victim. A precondition for this attack is
that the victim proxy does not support this extension since it
would not pass through overload control feedback otherwise.</t>
<t>A malicious SIP entity could gain an advantage by pretending to
support this specification but never reducing the amount of
traffic it forwards to the downstream neighbor. If its downstream
neighbor receives traffic from multiple sources which correctly
implement overload control, the malicious SIP entity would benefit
since all other sources to its downstream neighbor would reduce
load. </t>
<t><list>
<t>The solution to this problem depends on the
overload control method. For rate-based and window-based
overload control, it is very easy for a downstream entity to
monitor if the upstream neighbor throttles traffic forwarded as
directed. For percentage throttling this is not always obvious
since the load forwarded depends on the load received by the
upstream neighbor.</t>
</list></t>
</section>
<section anchor="sec:iana" title="IANA Considerations">
<t>This specification defines four new Via header parameters as
detailed below in the "Header Field Parameter and Parameter
Values" sub-registry as per the registry created by
<xref target="RFC3968"/>. The required information is:</t>
<figure><artwork>
Header Field Parameter Name Predefined Values Reference
__________________________________________________________
Via oc Yes RFCXXXX
Via oc-validity Yes RFCXXXX
Via oc-seq Yes RFCXXXX
Via oc-algo Yes RFCXXXX
RFC XXXX [NOTE TO RFC-EDITOR: Please replace with final RFC
number of this specification.]
</artwork></figure>
</section>
</middle>
<back>
<references title='Normative References'>
&rfc2119;
&rfc3261;
&rfc3263;
&rfc3968;
&rfc4412;
</references>
<references title='Informative References'>
&rfc5390;
&rfc5031;
&i-d.ietf-soc-overload-design;
&i-d.ietf-soc-load-control-event-package;
<reference anchor="I-D.soc-rate">
<front>
<title>Session Initiation Protocol (SIP) Rate Control</title>
<author initials="E." surname="Noel"><organization/></author>
<author initials="P." surname="Williams"><organization/></author>
<author initials="J." surname="Gunn"><organization/></author>
<date month="September" year="2011"/>
</front>
<seriesInfo name="Internet-Draft"
value="draft-noel-soc-overload-rate-control-00"/>
<format type="TXT"
target="http://www1.tools.ietf.org/html/draft-noel-soc-overload-rate-control.txt"/>
</reference>
</references>
<section title="Acknowledgements">
<t>Many thanks to Bruno Chatras, Keith Drage, Janet Gunn, Rich Terpstra,
Daryl Malas, R. Parthasarathi, Antoine Roly, Jonathan Rosenberg,
Charles Shen, Rahul Srivastava, Padma Valluri, Shaun Bharrat, and
Paul Kyzivat for their contributions to this specification.</t>
</section>
<section title="RFC5390 requirements" anchor="rfc5390-reqs">
<t><xref target="table-rfc5390"/> provides a summary how this
specification fulfills the requirements of <xref target="RFC5390"/>.
A more detailed view on how each requirements is fulfilled is
provided after the table.</t>
<texttable anchor="table-rfc5390">
<ttcol align="left">Requirement</ttcol>
<ttcol align="left">Meets requirement</ttcol>
<c>REQ 1</c> <c>Yes</c>
<c>REQ 2</c> <c>Yes</c>
<c>REQ 3</c> <c>Partially</c>
<c>REQ 4</c> <c>Partially</c>
<c>REQ 5</c> <c>Partially</c>
<c>REQ 6</c> <c>Not applicable</c>
<c>REQ 7</c> <c>Yes</c>
<c>REQ 8</c> <c>Partially</c>
<c>REQ 9</c> <c>Yes</c>
<c>REQ 10</c> <c>Yes</c>
<c>REQ 11</c> <c>Yes</c>
<c>REQ 12</c> <c>Yes</c>
<c>REQ 13</c> <c>Yes</c>
<c>REQ 14</c> <c>Yes</c>
<c>REQ 15</c> <c>Yes</c>
<c>REQ 16</c> <c>Yes</c>
<c>REQ 17</c> <c>Partially</c>
<c>REQ 18</c> <c>Yes</c>
<c>REQ 19</c> <c>Yes</c>
<c>REQ 20</c> <c>Yes</c>
<c>REQ 21</c> <c>Yes</c>
<c>REQ 22</c> <c>Yes</c>
<c>REQ 23</c> <c>Yes</c>
<postamble>Summary of meeting requirements in RFC5390</postamble>
</texttable>
<t>REQ 1: The overload mechanism shall strive to maintain the overall
useful throughput (taking into consideration the quality-of-service
needs of the using applications) of a SIP server at
reasonable levels, even when the incoming load on the network is
far in excess of its capacity. The overall throughput under load
is the ultimate measure of the value of an overload control mechanism.</t>
<t>Meeting REQ 1: Yes, the overload control mechanism allows an
overloaded SIP server to maintain a reasonable level of throughput as
it enters into congestion mode by requesting the upstream clients to
reduce traffic destined downstream.</t>
<t>REQ 2: When a single network element fails, goes into overload, or
suffers from reduced processing capacity, the mechanism should
strive to limit the impact of this on other elements in the
network. This helps to prevent a small-scale failure from becoming a
widespread outage.</t>
<t>Meeting REQ 2: Yes. When a SIP server enters overload mode, it
will request the upstream clients to throttle the traffic destined to it.
As a consequence of this, the overloaded SIP server will itself generate
proportionally less downstream traffic, thereby limiting the impact on
other elements in the network.</t>
<t>REQ 3: The mechanism should seek to minimize the amount of
configuration required in order to work. For example, it is
better to avoid needing to configure a server with its SIP message
throughput, as these kinds of quantities are hard to determine.</t>
<t>Meeting REQ 3: Partially. On the server side, the overload
condition is determined monitoring S (c.f., Section 4 of
<xref target="I-D.ietf-soc-overload-design"/>) and reporting a load
feedback F as a value to the "oc" parameter. On the client side,
a throttle T is applied to requests going downstream based on F.
This specification does not prescribe any value for S, nor a
particular value for F. The "oc-algo" parameter allows for automatic
convergence to a particular class of overload control algorithm.
There are suggested default values for the "oc-validity" parameter.</t>
<t>REQ 4: The mechanism must be capable of dealing with elements that
do not support it, so that a network can consist of a mix of
elements that do and don't support it. In other words, the
mechanism should not work only in environments where all elements
support it. It is reasonable to assume that it works better in
such environments, of course. Ideally, there should be
incremental improvements in overall network throughput as
increasing numbers of elements in the network support the mechanism.</t>
<t>Meeting REQ 4: Partially. The mechanism is designed to reduce congestion
when a pair of communicating entities support it. If a downstream
overloaded SIP server does not respond to a request in time, a SIP
client conformant to this specification will attempt to reduce traffic
destined towards the non-responsive server as outlined in
<xref target="sec:self-limiting"/>.</t>
<t>REQ 5: The mechanism should not assume that it will only be
deployed in environments with completely trusted elements. It
should seek to operate as effectively as possible in environments
where other elements are malicious; this includes preventing
malicious elements from obtaining more than a fair share of service.</t>
<t>Meeting REQ 5: Partially. Since overload control information is
shared between a pair of communicating entities, a confidential and
authenticated channel can be used for this communication. However,
if such a channel is not available, then the security ramifications
outlined in <xref target="sec:security"/> apply.</t>
<t>REQ 6: When overload is signaled by means of a specific message,
the message must clearly indicate that it is being sent because of
overload, as opposed to other, non overload-based failure
conditions. This requirement is meant to avoid some of the
problems that have arisen from the reuse of the 503 response code
for multiple purposes. Of course, overload is also signaled by
lack of response to requests. This requirement applies only to
explicit overload signals.</t>
<t>Meeting REQ 6: Not applicable. Overload control information is
signaled as part of the Via header and not in a new header.</t>
<t>REQ 7: The mechanism shall provide a way for an element to
throttle the amount of traffic it receives from an upstream
element. This throttling shall be graded so that it is not all-
or-nothing as with the current 503 mechanism. This recognizes the
fact that "overload" is not a binary state and that there are
degrees of overload.</t>
<t>Meeting REQ 7: Yes, please see <xref target="oc-use"/> and
<xref target="sec:respond-overload"/>.</t>
<t>REQ 8: The mechanism shall ensure that, when a request was not
processed successfully due to overload (or failure) of a
downstream element, the request will not be retried on another
element that is also overloaded or whose status is unknown. This
requirement derives from REQ 1.</t>
<t>Meeting REQ 8: Partially. A SIP client that has overload information
from multiple downstream servers will not retry the request
on another element. However, if a SIP client does not know the
overload status of a downstream server, it may send the request to
that server.</t>
<t>REQ 9: That a request has been rejected from an overloaded element
shall not unduly restrict the ability of that request to be
submitted to and processed by an element that is not overloaded.
This requirement derives from REQ 1.</t>
<t>Meeting REQ 9: Yes, a SIP client conformant to this specification
will send the request to a different element.</t>
<t>REQ 10: The mechanism should support servers that receive requests
from a large number of different upstream elements, where the set
of upstream elements is not enumerable.</t>
<t>Meeting REQ 10: Yes, there are no constraints on the number of
upstream clients.</t>
<t>REQ 11: The mechanism should support servers that receive requests
from a finite set of upstream elements, where the set of upstream
elements is enumerable.</t>
<t>Meeting REQ 11: Yes, there are no constraints on the number of
upstream clients.</t>
<t>REQ 12: The mechanism should work between servers in different domains.</t>
<t>Meeting REQ 12: Yes, there are no inherent limitations on using
overload control between domains.</t>
<t>REQ 13: The mechanism must not dictate a specific algorithm for
prioritizing the processing of work within a proxy during times of
overload. It must permit a proxy to prioritize requests based on
any local policy, so that certain ones (such as a call for
emergency services or a call with a specific value of the
Resource-Priority header field <xref target="RFC4412"/>) are given
preferential treatment, such as not being dropped, being given
additional retransmission, or being processed ahead of others.</t>
<t>Meeting REQ 13: Yes, please see <xref target="sec:respond-overload"/>.
</t>
<t>REQ 14: REQ 14: The mechanism should provide unambiguous directions to
clients on when they should retry a request and when they should
not. This especially applies to TCP connection establishment and
SIP registrations, in order to mitigate against avalanche restart.</t>
<t>Meeting REQ 14: Yes, <xref target="sec:self-limiting"/>
provides normative behavior on when to retry a request after repeated
timeouts and fatal transport errors resulting from communications with
a non-responsive downstream SIP server.</t>
<t>REQ 15: In cases where a network element fails, is so overloaded
that it cannot process messages, or cannot communicate due to a
network failure or network partition, it will not be able to
provide explicit indications of the nature of the failure or its
levels of congestion. The mechanism must properly function in
these cases.</t>
<t>Meeting REQ 15: Yes, <xref target="sec:self-limiting"/>
provides normative behavior on when to retry a request after repeated
timeouts and fatal transport errors resulting from communications with
a non-responsive downstream SIP server.</t>
<t>REQ 16: The mechanism should attempt to minimize the overhead of
the overload control messaging.</t>
<t>Meeting REQ 16: Yes, overload control messages are sent in the
topmost Via header, which is always processed by the SIP elements.</t>
<t>REQ 17: The overload mechanism must not provide an avenue for
malicious attack, including DoS and DDoS attacks.</t>
<t>Meeting REQ 17: Partially. Since overload control information is
shared between a pair of communicating entities, a confidential and
authenticated channel can be used for this communication. However,
if such a channel is not available, then the security ramifications
outlined in <xref target="sec:security"/> apply.</t>
<t>REQ 18: The overload mechanism should be unambiguous about whether
a load indication applies to a specific IP address, host, or URI,
so that an upstream element can determine the load of the entity
to which a request is to be sent.</t>
<t>Meeting REQ 18: Yes, please see discussion in <xref target="oc-use"/>.</t>
<t>REQ 19: The specification for the overload mechanism should give
guidance on which message types might be desirable to process over
others during times of overload, based on SIP-specific
considerations. For example, it may be more beneficial to process
a SUBSCRIBE refresh with Expires of zero than a SUBSCRIBE refresh
with a non-zero expiration (since the former reduces the overall
amount of load on the element), or to process re-INVITEs over new INVITEs.</t>
<t>Meeting REQ 19: Yes, please see <xref target="sec:respond-overload"/>.</t>
<t>REQ 20: In a mixed environment of elements that do and do not
implement the overload mechanism, no disproportionate benefit
shall accrue to the users or operators of the elements that do not
implement the mechanism.</t>
<t>Meeting REQ 20: Yes, an element that does not implement overload
control does not receive any measure of extra benefit.</t>
<t>REQ 21: The overload mechanism should ensure that the system
remains stable. When the offered load drops from above the
overall capacity of the network to below the overall capacity, the
throughput should stabilize and become equal to the offered load.</t>
<t>Meeting REQ 21: Yes, the overload control mechanism described in
this draft ensures the stability of the system.</t>
<t>REQ 22: It must be possible to disable the reporting of load
information towards upstream targets based on the identity of
those targets. This allows a domain administrator who considers
the load of their elements to be sensitive information, to
restrict access to that information. Of course, in such cases,
there is no expectation that the overload mechanism itself will
help prevent overload from that upstream target.</t>
<t>Meeting REQ 22: Yes, an operator of a SIP server can configure
the SIP server to only report overload control information for
requests received over a confidential channel, for example. However,
note that this requirement is in conflict with REQ 3, as it
introduces a modicum of extra configuration.</t>
<t>REQ 23: It must be possible for the overload mechanism to work in
cases where there is a load balancer in front of a farm of proxies.</t>
<t>Meeting REQ 23: Yes. Depending on the type of load balancer,
this requirement is met. A load balancer fronting
a farm of SIP proxies could be a SIP-aware load balancer or one that
is not SIP-aware. If the load balancer is SIP-aware, it can make
conscious decisions on throttling outgoing traffic towards the
individual server in the farm based on the overload control parameters
returned by the server. On the other hand, if the load balancer is
not SIP-aware, then there are other strategies to perform overload
control. Section 6 of <xref target="I-D.ietf-soc-overload-design"/>
documents some of these strategies in more detail (see discussion related
to Figure 3(a) in Section 6).</t>
<!--
For list discussion on this, see
http://www.ietf.org/mail-archive/web/sip-overload/current/msg00531.html
http://www.ietf.org/mail-archive/web/sip-overload/current/msg00533.html
-->
</section> <!-- rfc5390-reqs -->
</back>
</rfc>
| PAFTECH AB 2003-2026 | 2026-04-24 08:18:25 |