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{{DISPLAYTITLE: Brief Tutorial of the SIP Signaling and SDP Media Protocols }}
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The VoIP Switch administrators, operators and supporters find here information about the Session Initiation Protocol SIP and Session Description Protocol SDP.


Basic knowhow about connection signalization with '''Session Initiation Protocol SIP''':
:* [[ {{NAMESPACE}}:support_voip_protocol#SupportSipBasics        | Basics about SIP Protocol ]]
:* [[ {{NAMESPACE}}:support_voip_protocol#SupportSipResponseCodes | List of most important SIP Response Codes ]]


Basic knowhow about media (speech) transmission with '''Session Description Protocol SDP''':
:* [[ {{NAMESPACE}}:support_voip_protocol#SupportSdpBasics        | Basics about SDP Protocol ]]
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{{ToTop | SupportSipKnowhow }} <!------------------------------------------------------------------>
= Knowhow Connection Signaling with "Session Initiation Protocol SIP" =

The '''Session Initiation Protocol SIP''' is a communications protocol for signaling and controlling multimedia communication sessions. One of the most common applications of SIP is in Internet telephony for voice and video calls.

For an extended overview of the SIP protocol visit:
: <tt>[https://en.wikipedia.org/wiki/Session_Initiation_Protocol Wikipedia: Session Initiation Protocol SIP]</tt>



{{ToTop | SupportSipBasics }} <!------------------------------------------------------------------->
== Basics: Session Session Protocol SIP ==

Example of a "SIP dialog" with the minimal needed messages for a connection setup or connection renegotiation:
[[file:support_sip_connection_setup.png |575px|left|frameless|link=| SIP Connection Establishing ]]
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Example of a "SIP dialog" with the minimal needed messages for a connection release:
[[file:support_sip_connection_release.png |575px|left|frameless|link=| SIP Connection Release ]]
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{{ToTop | SupportSipFlow }} <!--------------------------------------------------------------------->
== Examples: SIP Signaling Flows ==

Example of a regular outgoing call into the PSTN:
[[file:support_sip_flow_pstn_outgoing_regular.png |680px|left|frameless|link=| SIP Flow PSTN Outgoing ]]
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Example of a regular incoming call from the PSTN:
[[file:support_sip_flow_pstn_incoming_regular.png |575px|left|frameless|link=| SIP Flow PSTN Incoming  ]]
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Example of an outgoing call into the PSTN with three exceptional signaling situations:
:# The PSTN Gateway 1 doesn't respond so the VoIP Switch has to re-route to the PSTN Gateway 2
:# The telephone on side A offers an invalid "Session Time" value which is refused by the PSTN Gateway 2. The telephone on side A has to do a reINVITE with an acceptable "Session Time" value.
:# End point B is busy.
[[file:support_sip_flow_pstn_outgoing_gw_failover_regular.png |750px|left|frameless|link=| SIP Flow GW Fail Over ]]
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Example of a connection where the VoIP Switch checks the presence of the end points with OPTION messages. The VoIP Switch would release the connection if one end point doesn't respond with "200 OK":
[[file:support_sip_flow_pstn_outgoing_options_regular.png |575px|left|frameless|link=| SIP Flow PSTN Outgoing Options ]]
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{{ToTop | SupportSipResponseCodes }} <!------------------------------------------------------------>
== SIP Response Codes ==

A list of SIP response codes and their meaning can be found here:
:<tt>[https://en.wikipedia.org/wiki/List_of_SIP_response_codes Wikipedia: List of SIP Response Codes ]</tt>




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=== Most Important 1xx—Provisional Responses ===

'''100 Trying'''<br>
Extended search being performed may take a significant time so a forking proxy must send a 100 Trying response.

'''180 Ringing'''<br>
Destination user agent received INVITE, and is alerting user of call.

'''183 Session in Progress'''<br>
This response may be used to send extra information for a call which is still being set up.

<!--
'''181 Call is Being Forwarded'''<br>
Servers can optionally send this response to indicate a call is being forwarded.

'''182 Queued'''<br>
Indicates that the destination was temporarily unavailable, so the server has queued the call until the destination is available. A server may send multiple 182 responses to update progress of the queue.

'''199 Early Dialog Terminated'''<br>
Can be used by User Agent Server to indicate to upstream SIP entities (including the User Agent Client (UAC)) that an early dialog has been terminated.
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=== Most Important 2xx—Successful Responses ===

'''200 OK'''<br>
Indicates the request was successful.

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'''202 Accepted'''<br>
Indicates that the request has been accepted for processing, but the processing has not been completed.

'''204 No Notification'''<br>
Indicates the request was successful, but the corresponding response will not be received.
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=== Most Important 3xx—Redirection Responses ===

'''302 Moved Temporarily'''<br>
The client should try at the address in the Contact field. If an Expires field is present, the client may cache the result for that period of time.

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'''300 Multiple Choices'''<br>
The address resolved to one of several options for the user or client to choose between, which are listed in the message body or the message's Contact fields.

'''301 Moved Permanently'''<br>
The original Request-URI is no longer valid, the new address is given in the Contact header field, and the client should update any records of the original Request-URI with the new value.

'''305 Use Proxy'''<br>
The Contact field details a proxy that must be used to access the requested destination.

'''380 Alternative Service'''<br>
The call failed, but alternatives are detailed in the message body.
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=== Most Important 4xx—Client Failure Responses ===

'''400 Bad Request'''<br>
The request could not be understood due to malformed syntax.

'''401 Unauthorized'''<br>
The request requires user authentication. This response is issued by UASs and registrars.

'''403 Forbidden'''<br>
The server understood the request, but is refusing to fulfil it.

'''404 Not Found'''<br>
The server has definitive information that the user does not exist at the domain specified in the Request-URI. This status is also returned if the domain in the Request-URI does not match any of the domains handled by the recipient of the request.

'''406 Not Acceptable'''<br>
The resource identified by the request is only capable of generating response entities that have content characteristics but not acceptable according to the Accept header field sent in the request.

'''408 Request Timeout'''<br>
Couldn't find the user in time. The server could not produce a response within a suitable amount of time, for example, if it could not determine the location of the user in time. The client MAY repeat the request without modifications at any later time.

'''410 Gone'''<br>
The user existed once, but is not available here any more.

'''480 Temporarily Unavailable'''<br>
Callee currently unavailable.

'''486 Busy Here'''<br>
Callee is busy.

'''487 Request Terminated'''<br>
Request has terminated by bye or cancel.

'''488 Not Acceptable Here'''<br>
Some aspect of the session description or the Request-URI is not acceptable.

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'''402 Payment Required'''<br>
Reserved for future use.

'''405 Method Not Allowed'''<br>
The method specified in the Request-Line is understood, but not allowed for the address identified by the Request-URI.[1]:

'''407 Proxy Authentication Required'''<br>
The request requires user authentication. This response is issued by proxys.

'''409 Conflict'''<br>
User already registered.

'''411 Length Required'''<br>
The server will not accept the request without a valid Content-Length.

'''412 Conditional Request Failed'''<br>
The given precondition has not been met.

'''413 Request Entity Too Large'''<br>
Request body too large.

'''414 Request-URI Too Long'''<br>
The server is refusing to service the request because the Request-URI is longer than the server is willing to interpret.

'''415 Unsupported Media Type'''<br>
Request body in a format not supported.

'''416 Unsupported URI Scheme'''<br>
Request-URI is unknown to the server.

'''417 Unknown Resource-Priority'''<br>
There was a resource-priority option tag, but no Resource-Priority header.

'''420 Bad Extension'''<br>
Bad SIP Protocol Extension used, not understood by the server.

'''421 Extension Required'''<br>
The server needs a specific extension not listed in the Supported header.

'''422 Session Interval Too Small'''<br>
The received request contains a Session-Expires header field with a duration below the minimum timer.

'''423 Interval Too Brief'''<br>
Expiration time of the resource is too short.

'''424 Bad Location Information'''<br>
The request's location content was malformed or otherwise unsatisfactory.

'''428 Use Identity Header'''<br>
The server policy requires an Identity header, and one has not been provided.

'''429 Provide Referrer Identity'''<br>
The server did not receive a valid Referred-By token on the request.

'''430 Flow Failed'''<br>
A specific flow to a user agent has failed, although other flows may succeed. This response is intended for use between proxy devices, and should not be seen by an endpoint (and if it is seen by one, should be treated as a 400 Bad Request response).

'''433 Anonymity Disallowed'''<br>
The request has been rejected because it was anonymous.

'''436 Bad Identity-Info'''<br>
The request has an Identity-Info header, and the URI scheme in that header cannot be dereferenced.

'''437 Unsupported Certificate'''<br>
The server was unable to validate a certificate for the domain that signed the request.

'''438 Invalid Identity Header'''<br>
The server obtained a valid certificate that the request claimed was used to sign the request, but was unable to verify that signature.

'''439 First Hop Lacks Outbound Support'''<br>
The first outbound proxy the user is attempting to register through does not support the "outbound" feature of RFC 5626, although the registrar does.

'''470 Consent Needed'''<br>
The source of the request did not have the permission of the recipient to make such a request.

'''481 Call/Transaction Does Not Exist'''<br>
Server received a request that does not match any dialog or transaction.

'''482 Loop Detected.'''<br>
Server has detected a loop.

'''483 Too Many Hops'''<br>
Max-Forwards header has reached the value '0'.

'''484 Address Incomplete'''<br>
Request-URI incomplete.

'''485 Ambiguous'''<br>
Request-URI is ambiguous.

'''489 Bad Event'''<br>
The server did not understand an event package specified in an Event header field.

'''491 Request Pending'''<br>
Server has some pending request from the same dialog.

'''493 Undecipherable'''<br>
Request contains an encrypted MIME body, which recipient can not decrypt.[1]:§21.4.28

'''494 Security Agreement Required'''<br>
The server has received a request that requires a negotiated security mechanism, and the response contains a list of suitable security mechanisms for the requester to choose between or a digest authentication challenge.
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=== Most Important 5xx—Server Failure Responses ===


'''503 Service Unavailable'''<br>
The server is undergoing maintenance or is temporarily overloaded and so cannot process the request. A "Retry-After" header field may specify when the client may reattempt its request.

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'''500 Server Internal Error'''<br>
The server could not fulfill the request due to some unexpected condition.

'''501 Not Implemented'''<br>
The server does not have the ability to fulfill the request, such as because it does not recognize the request method. (Compare with 405 Method Not Allowed, where the server recognizes the method but does not allow or support it.)

'''502 Bad Gateway'''<br>
The server is acting as a gateway or proxy, and received an invalid response from a downstream server while attempting to fulfill the request.

'''504 Server Time-out'''<br>
The server attempted to access another server in attempting to process the request, and did not receive a prompt response.

'''505 Version Not Supported'''<br>
The SIP protocol version in the request is not supported by the server.

'''513 Message Too Large'''<br>
The request message length is longer than the server can process.

'''580 Precondition Failure'''<br>
The server is unable or unwilling to meet some constraints specified in the offer.
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=== Most Important 6xx—Global Failure Responses ===

'''603 Decline'''<br>
The destination does not wish to participate in the call, or cannot do so, and additionally the destination knows there are no alternative destinations (such as a voicemail server) willing to accept the call.

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'''600 Busy Everywhere'''<br>
All possible destinations are busy. Unlike the 486 response, this response indicates the destination knows there are no alternative destinations (such as a voicemail server) able to accept the call.

'''604 Does Not Exist Anywhere'''<br>
The server has authoritative information that the requested user does not exist anywhere.

'''606 Not Acceptable'''<br>
The user's agent was contacted successfully but some aspects of the session description such as the requested media, bandwidth, or addressing style were not acceptable.
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{{ToTop | SupportSdpKnowhow }} <!------------------------------------------------------------------>
= Knowhow Media Stream Signaling with "Session Description Protocol SDP" =

The '''Session Description Protocol SDP''' describes how during a connection setup the end points negotiate the parameters of this exchange as session announcement, session invitation, and parameter. SDP does not deliver media itself but is used between end points for negotiation of media type, format, and all associated properties for voice, Fax, DTMF, bit transparent data etc..

For an extended overview of the SDP protocol visit [https://en.wikipedia.org/wiki/Session_Initiation_Protocol Wikipedia].


{{Note|
The VoIP Switch doesn't interfere in the SDP negotiation of the end points!
There may be exceptions for certain Customer Premises Equipment CPE devices where interoperation problems are known. Check with the VoIP switch administrator which CPE devices are known with SDP manipulations by the VoIP switch.
}}




{{ToTop | SupportSdpBasics }} <!------------------------------------------------------------------->
== Basics: Session Description Protocol SDP ==

The SDP is embedded in the SIP messages during connection setup or connection renegotiation:
[[file:support_sip_connection_setup_with_sdp.png |575px|left|frameless|link=| SIP with SDP]]
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The following SDP properties and parameters are important for supporting customer problems:
[[file:support_sdp_basic.png                     |730px|left|frameless|link=| SDP Basic Properties and Parameters ]]
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Example of a SDP offer from the calling side A:
[[file:support_sdp_offer.png                     |730px|left|frameless|link=| SDP Offer Calling Side A ]]
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Example of a SDP answer from the called side B:
[[file:support_sdp_answer.png                    |730px|left|frameless|link=| SDP Answer Called Side B ]]
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Example of a SDP offer for a Fax transfer with T.38 from the calling side A:
[[file:support_sdp_t38.png                      |730px|left|frameless|link=| SDP Offer Calling Side A with T.38 ]]
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Interpretation of the "Media Attributes":
{{Table_Start}}
{{Table_Header | columns=4 | width1=50 | width2=150 | width3=250 | width4=550
| header1= Index | header2= Type | header3= Attribute | header4= Remark }}

{{Table_Line   | columns=4 | line=1
| col1= 0      | col2= PCMU  | col3= ISDN G.711&micro;law | col4= Very good quality VoIP codec }}

{{Table_Line   | columns=4  | line=2
| col1= 8      | col2= PCMA  | col3= ISDN G.711alaw | col4= Very good quality VoIP codec }}

{{Table_Line   | columns=4 | line=3
| col1= 2      | col2= G.726-32 | col3= &nbsp; | col4= Good quality VoIP codec }}

{{Table_Line   | columns=4 | line=4
| col1= 18     | col2= G.729 | col3= &nbsp; | col4= Low quality VoIP codec }}

{{Table_Line   | columns=4 | line=5
| col1=125     | col2= x-clear-channel | col3= data service bit transparent | col4= Echo canceling will be switched off and the data bit by bit transferred }}

{{Table_Line   | columns=4 | line=6
| col1= 101    | col2= telephone-event  | col3= DTMF, RFC 2833 | col4= DTMF will not be transferred inband but as RTP event according RFC 2833 }}

{{Table_Line   | columns=4 | line=7
| col1= 18     | col2= annexb=0  | col3= Special information for codec with index 18 | col4= Special directive for codec G.729 }}

{{Table_Line   | columns=4 | line=8
| col1= 101    | col2= 0-16  | col3= Special information for for telephone-event with index 101
| col4=
0-15 : DTMF character 0-9, *,#, A,B,C,D <br>
0-16 : DTMF character 0-9, *,#, A,B,C,D, Flash
}}
{{Table_End}}




{{ToTop | SupportRtpBasics }} <!------------------------------------------------------------------->

== Basics: RTP/RTCP ==

The Real Time Protocol RTP is used to transfer media data, e.g. speech in VoIP based telephony.

The Real Time Control Protocol RTPC transfers periodically statistical media data between the peers of a connection.

If RTP packets are lost, delayed or jitter then we speak of a Quality of Service QoS problem. For the support it is of interest to know if the number of transferred packets between the peers of a connection and if the numbers in the receive and send paths are reasonable equal, if packets were lost on call leg etc. With these statistical media information it can be possible to identify a path or transfer direction were QoS problems occure.

{{Note|
The media stream must be proxied via the MediaServer of the VoIP Switch in order to compute statistical numbers of a connection.
}}


The Aarenet VoIP Switch supports RTP/RTCP statistic data collection of a connection. How they can be obtained is described in article "Manual of the Aarenet VoIP Switch Support Tools", [[{{NAMESPACE}}:support_tools#SupportToolConfigCenterCallData | chapter "The ConfigCenter Call Data" ]]


Overview of "RTP/RTCP" information collection:
[[file:support_connection_rtp_rtcp_statistics_e.png |1000px|left|frameless| Dialog: "RTP/RTCP" ]]
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Details of "RTP/RTCP" information collection:
[[file:support_connection_rtp_rtcp_statistics_details_e.png |1000px|left|frameless| Dialog: "RTP/RTCP Details" ]]
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