[Frage] Cisco 9971 - Webradio streaming

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Hallo zusammen,

dank die Infos hier habe ich mein Cisco 9971 erfolgreich an mein Fritzbox 7580 angeschlossen. Jetzt möchte ich versuchen streaming Dienste wie WebRadio anzubinden. ich habe zwar eine Erwähnung hier gefunden, bin aber unschlüssig wie das funktioniert.

Laut das Cisco Programming Guide unter "URIs to Control RTP Streaming" steht folgendes zum Thema RTPx:

RTPRx

The RTPRx URI instructs the phone to receive a Unicast RTP stream or to stop receiving Unicast or Multicast RTP streams.

URI Formats:
RTPRx :i: p:v
RTPRx:Stop

Where

i = the IP Address from which the stream is coming.

p = the UDP port on which to receive the RTP stream. Ensure that this is an even port number within the decimal range of 20480 to 32768. If no port is specified, the phone chooses a port and returns it when initiated by a push request.

Stop = the parameter that will stop any active RTP stream from being received on channel one

v = the optional volume setting that controls the volume of stream playout. The supplied value is a percentage of the maximum volume level of the device and must be in the range 0-100. The phone converts the specified percentage into the closest device-supported volume level setting and uses it. After the initial volume level gets set and the stream starts, you can manually change the volume level as needed. If the optional volume parameter does not get included, the current volume setting on the phone gets used as the default.

Soweit so klar, aber die meisten Webradios haben mehrere Streams an ihre IP Adresse und ein URL jeweils mit .mp3 am Ende. (z.B. Antenne Bayern). Hat jemand das möglicherweise selber versucht und kann mir anhand von ein Beispiel die Anwendung und Syntax zeigen?

Danke!
 
Zuletzt bearbeitet:
Moin


Nicht ohne eine Transkodierung.
Denn laut deiner Verlinkung muss der Stream ...

The supported format of the RTP stream is as follows:
blank.gif
The codec is G.711 mu-Law.
blank.gif
The packet size is 20 ms.

..."Telefonqualität" haben.
 
Ich habe folgendes bei Cisco gefunden:
9971 RTP Streaming API and Speakerphone
Subject: RE: 9971 RTP Streaming API and Speakerphone
Replied by: JAMES DEPHILLIP II on 12-12-2012 09:57:37 AM
Say I use the following XML...This works great on a 7900 Series even if speakerphone is disabled it will start receiving the multicast stream through the speaker of the IP Phone. The 9971 however has two issues. If speakerphone is disabled it will not work. It will work if I take the handset off hook but I can not get it to operate through the speaker. Also the 9971 does not work with onStopped. Keep in mind that both of these work on a 7965 without a single issue.

<startMedia>
<mediaStream onStopped=”Notify:http:10.1.2.15:80:demo/log.php”>
<type>audio</type>
<codec>G.722</codec>
<mode>receive</mode>
<address>239.1.2.3</address>
<port>20480</port>
</mediaStream>
</startMedia>
----------------------------------------------------------------------------------
Hi James,
The 9971 will work using the RTP Streaming URIs:

<CiscoIPPhoneExecute>
<ExecuteItem URL=“RTPRx:1.2.3.4:20480”/>
<ExecuteItem URL=“RTPTx:4.3.2.1:20480”/>
</CiscoIPPhoneExecute>

---------------------------------------------------------------------------------
Replied by: JAMES DEPHILLIP II on 12-12-2012 09:22:58 PM
I can not use the legacy URI's because they do not support G.722. Keep in mind that the RTP API works fine unless speakerphone is disabled [...]

scheinbar geht es (oder müsste es gehen) mit G.722. Der Verfasser möchte das es mit sein Lautsprecher auf Disabled geht, was bei mir nicht der Fall ist.
Nicht ohne eine Transkodierung.
Denn laut deiner Verlinkung muss der Stream [...] "Telefonqualität" haben.
Ich teste das erstmal mit ein Stream von VLC Media Player mittels RTP und das <CiscoIPPhoneExecute> Tag. danach schaue ich ob ich mittels API irgendeine Verbesserung zum laufen bringen kann.

-- Zusammenführung Doppelpost by stoney

Bzgl. das RTP API hier nur vollständigkeitshalber die Information von Cisco:

Cisco Unified IP Phone Services Application Development Notes for Cisco Unified Communications Manager and Multiplatform Phones
RTP Streaming API

This XML-based RTP Streaming API allows applications to initiate and observe RTP audio streams. This API extends capabilities beyond the legacy RTP streaming URIs by providing support for stream start and stop event listeners and the ability to specify other extended stream attributes, such as codec type.


note.gif
Note
Support for the RTP Streaming API requires an updated XML Parser.

The Multiplatform phones doe not support the RTP Streaming API.

The event handlers typically use the standard Notification framework, but they can also invoke most other URIs, with the exception of HTTP URLs.
Related References
Updated XML Parser and Schema Enforcement
Notify

Interaction Rules with Legacy RTP URI Streams

The RTP Streaming API allows a full-duplex stream (mode=sendReceive) to be set up as a single stream request, which simplifies the usage of the API. However, in some cases, this API creates some interoperability issues with the legacy RTP URIs because the legacy RTP URIs send and receive streams separately. The interaction rules between legacy RTP URI streams and the new RTP Streaming API are:
  • If an RTP Stop URI is invoked, and an RTP Streaming API stream is currently streaming in that same direction, then the entire RTP Streaming API stream is stopped.

    For example, if a full-duplex stream is set up through the RTP Streaming API (mode=sendReceive) and then an RTPTx:Stop URI is invoked, the stream will be stopped in both the send and receive directions (and the onStopped event handler will be called, if present).
  • If the stopMedia request (from the RTP Streaming API) does not specify a stream ID, then the request will stop all services RTP streams, in any direction (send or receive) and of any type (multicast and unicast). This allows applications using the RTP Streaming API to stop media streams which may have been started by the legacy RTP URIs or by other applications for which a stream ID is not known.

Error Schema
XML:
<?xml version="1.0" encoding="UTF-8"?>
<xs:schema xmlns:xs="http://www.w3.org/2001/XMLSchema" elementFormDefault="qualified" attributeFormDefault="unqualified">
  <xs:element name="errorResponse">
    <xs:complexType>
      <xs:all>
        <xs:element name="type">
          <xs:simpleType>
            <xs:restriction base="xs:string">
              <xs:enumeration value="InvalidURL"/>
              <xs:enumeration value="InvalidResource"/>
              <xs:enumeration value="InvalidResourceID"/>
              <xs:enumeration value="UnavailableResource"/>
              <xs:enumeration value="InvalidXML"/>
            </xs:restriction>
          </xs:simpleType>
        </xs:element>
        <xs:element name="data" nillable="true">
          <xs:simpleType>
            <xs:restriction base="xs:string"/>
          </xs:simpleType>
        </xs:element>
      </xs:all>
    </xs:complexType>
  </xs:element>
</xs:schema>

RTP Streaming API Examples
The following examples show how to work with the RTP Streaming API.

Start Media Example

Request
Code:
HTTP POST /CGI/Execute
<startMedia>
<mediaStream
onStopped=”Notify:http:server:80:path/page”
receiveVolume=”50”>
<type>audio</type>
<codec>G.729</codec>
<mode>sendReceive</mode>
<address>239.1.2.3</address>
<port>20480</port>
</mediaStream>
</startMedia>

Response
Code:
HTTP200 OK
<mediaStream id=”abc123”/>


Stop Media Example

Request
Code:
HTTP POST CGI/Execute
<stopMedia>
<mediaStream id=”abc123”/>
</stopMedia>

Response
Code:
HTTP 200 OK


If the user terminates the media stream by placing the active audio path on-hook, the following notification is sent:
Code:
HTTP POST /server/path/page
DATA=<notifyMediaEvent type=”stopped” origin=”user”>
    <mediaStream id=”abc123”/>
</notifyMediaEvent>
 
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