Support voip protocol - Revision history

Anadm at 13:56, 10 October 2017

2017-10-10T13:56:45Z

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Anadm at 14:41, 6 October 2017

2017-10-06T14:41:36Z

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Anadm: /* Basics: Session Description Protocol SDP */

2017-10-06T07:50:16Z

‎Basics: Session Description Protocol SDP

Anadm: /* Basics: RTP/RTCP */

2017-09-18T08:17:29Z

‎Basics: RTP/RTCP

Anadm at 15:15, 6 September 2017

2017-09-06T15:15:29Z

Anadm at 13:19, 5 September 2017

2017-09-05T13:19:24Z

Anadm at 15:09, 4 September 2017

2017-09-04T15:09:09Z

Anadm at 14:24, 4 September 2017

2017-09-04T14:24:13Z

Anadm at 14:29, 31 August 2017

2017-08-31T14:29:13Z

Anadm at 09:43, 30 August 2017

2017-08-30T09:43:00Z

@@ Line 378: / Line 378: @@
 <br clear=all>
+<!--
 Example of a SDP answer from the called side B:
 [[file:support_sdp_answer.png                    |730px|left|frameless|link=| SDP Answer Called Side B ]]
 <br clear=all>
+-->
 Example of a SDP offer for a Fax transfer with T.38 from the calling side A:
@@ Line 398: / Line 398: @@
 {{Table_Line   | columns=4  | line=2
-| col1= 1      | col2= PCMA  | col3= ISDN G.711alaw | col4= Very good quality VoIP codec }}
+| col1= 8      | col2= PCMA  | col3= ISDN G.711alaw | col4= Very good quality VoIP codec }}
 {{Table_Line   | columns=4 | line=3
@@ Line 427: / Line 427: @@
 {{ToTop | SupportRtpBasics }} <!------------------------------------------------------------------->
 == Basics: RTP/RTCP ==

@@ Line 444: / Line 444: @@
 Overview of "RTP/RTCP" information collection:
-[[file:support_connection_rtp_rtcp_statistics_e.png |750px|left|frameless| Dialog: "RTP/RTCP" ]]
+[[file:support_connection_rtp_rtcp_statistics_e.png |1000px|left|frameless| Dialog: "RTP/RTCP" ]]
 <br clear=all>
 Details of "RTP/RTCP" information collection:
-[[file:support_connection_rtp_rtcp_statistics_details_e.png |750px|left|frameless| Dialog: "RTP/RTCP Details" ]]
+[[file:support_connection_rtp_rtcp_statistics_details_e.png |1000px|left|frameless| Dialog: "RTP/RTCP Details" ]]
 <br clear=all>

@@ Line 107: / Line 107: @@
 '''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.
+-->
@@ Line 123: / Line 123: @@
 '''204 No Notification'''<br>
 Indicates the request was successful, but the corresponding response will not be received.
+-->
@@ Line 146: / Line 146: @@
 The call failed, but alternatives are detailed in the message body.
 -->

@@ Line 3: / Line 3: @@
 {{Page_Menu_List_Help_Support}}
 <!-- FOOTER LEFT "© Aarenet AG 2017" --> <!-- PDF Creation Directive ------------------------------>
@@ Line 24: / Line 23: @@
 {{ToTop | SupportSipKnowhow }} <!------------------------------------------------------------------>
 = Knowhow Connection Signaling with "Session Initiation Protocol SIP" =

@@ Line 429: / Line 429: @@
 {{ToTop | SupportRtpBasics }} <!------------------------------------------------------------------->
 == Basics: RTP/RTCP ==
 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" ]]
@@ Line 434: / Line 445: @@
 Overview of "RTP/RTCP" information collection:
-[[file:support_connection_rtp_rtcp_statistics_e.png |750px|left|frameless|link=| Dialog: "RTP/RTCP" ]]
+[[file:support_connection_rtp_rtcp_statistics_e.png |750px|left|frameless| Dialog: "RTP/RTCP" ]]
 <br clear=all>
 Details of "RTP/RTCP" information collection:
-[[file:support_connection_rtp_rtcp_statistics_details_e.png |750px|left|frameless|link=| Dialog: "RTP/RTCP Details" ]]
+[[file:support_connection_rtp_rtcp_statistics_details_e.png |750px|left|frameless| Dialog: "RTP/RTCP Details" ]]
 <br clear=all>

@@ Line 25: / Line 25: @@
 {{ToTop | SupportSipKnowhow }} <!------------------------------------------------------------------>
-= Knowhow Connection Signaling with "Session Initiation Protocol SIP" {{Help_Status |  | &nbsp }} =
+= 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.
@@ Line 35: / Line 35: @@
 {{ToTop | SupportSipBasics }} <!------------------------------------------------------------------->
-== Basics: Session Session Protocol SIP {{Help_Status |  | &nbsp }} ==
+== Basics: Session Session Protocol SIP ==
 Example of a "SIP dialog" with the minimal needed messages for a connection setup or connection renegotiation:
@@ Line 50: / Line 50: @@
 {{ToTop | SupportSipFlow }} <!--------------------------------------------------------------------->
-== Examples: SIP Signaling Flows {{Help_Status |  | &nbsp }} ==
+== Examples: SIP Signaling Flows ==
 Example of a regular outgoing call into the PSTN:
@@ Line 347: / Line 347: @@
 {{ToTop | SupportSdpKnowhow }} <!------------------------------------------------------------------>
-= Knowhow Media Stream Signaling with "Session Description Protocol SDP" {{Help_Status |  | &nbsp }} =
+= 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..
@@ Line 363: / Line 363: @@
 {{ToTop | SupportSdpBasics }} <!------------------------------------------------------------------->
-== Basics: Session Description Protocol SDP {{Help_Status |  | &nbsp }} ==
+== Basics: Session Description Protocol SDP ==
 The SDP is embedded in the SIP messages during connection setup or connection renegotiation:
@@ Line 424: / Line 424: @@
 {{Table_End}}

@@ Line 1: / Line 1: @@
-{{DISPLAYTITLE: Knowhow VoIP Protocols }}
+{{DISPLAYTITLE: Brief Tutorial of the SIP Signaling and SDP Media Protocols }}
 {{PAGE_HEADER}}
 {{Page_Menu_List_Help_Support}}
@@ Line 8: / Line 8: @@
 {{Page_Introduction_Title}}
 <!-- Transclusion : Intro  Begin ------------------------------------------------------------------><section begin=intro />
-The VoIP Switch Administrators, Operators, Supporters find here information about the Session Initiation Protocol SIP and Session Description Protocol SDP.
+The VoIP Switch administrators, operators and supporters find here information about the Session Initiation Protocol SIP and Session Description Protocol SDP.

@@ Line 66: / Line 66: @@
 :# 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 |575px|left|frameless|link=| SIP Flow GW Fail Over ]]
+[[file:support_sip_flow_pstn_outgoing_gw_failover_regular.png |750px|left|frameless|link=| SIP Flow GW Fail Over ]]
 <br clear=all>
@@ Line 396: / Line 396: @@
 {{Table_Line   | columns=4 | line=1
-| col1= &nbsp; | col2= G.726-32 | col3= &nbsp; | col4= Good quality VoIP codec }}
+| col1= 0      | col2= PCMU  | col3= ISDN G.711&micro;law | col4= Very good quality VoIP codec }}
-{{Table_Line   | columns=4 | line=2
+{{Table_Line   | columns=4  | line=2
-| col1= 18     | col2= G.729 | col3= &nbsp; | col4= Low quality VoIP codec }}
+| col1= 1      | col2= PCMA  | col3= ISDN G.711alaw | col4= Very good quality VoIP codec }}
-{{Table_Line   | columns=4  | line=3
+{{Table_Line   | columns=4 | line=3
-| col1= &nbsp; | col2= PCMA  | col3= ISDN G.711alaw | col4= Very good quality VoIP codec }}
+| col1= 2      | col2= G.726-32 | col3= &nbsp; | col4= Good quality VoIP codec }}
 {{Table_Line   | columns=4 | line=4
-| col1= &nbsp; | col2= PCMU  | col3= ISDN G.711&micro;law | col4= Very good quality VoIP codec }}
+| col1= 18     | col2= G.729 | col3= &nbsp; | col4= Low quality VoIP codec }}
 {{Table_Line   | columns=4 | line=5
@@ Line 419: / Line 419: @@
 | col1= 101    | col2= 0-16  | col3= Special information for for telephone-event with index 101
 | col4=
--15 : DTMF Zeichen 0-9, *,#, A,B,C,D <br>
+-15 : DTMF character 0-9, *,#, A,B,C,D <br>
--16 : DTMF Zeichen 0-9, *,#, A,B,C,D, Flash
+-16 : DTMF character 0-9, *,#, A,B,C,D, Flash
 }}
 {{Table_End}}