Hello,

I need some help on using RTP Streaming API with different codec settings other than G.711.
I am able to use the G.711 codec successfully but I am not able to work with other codec types.
I am using Sun JMF(Java Media Framework) studio and VoiceAge G.729 java plug-in for testing. JMF comes default with G.711/ULAW, G.723 and GSM codecs.

When I use the send mode I can listen the audio stream from JMF studio successfully for G.729 codec.

But when I send the stream from JMF selecting G.729/RTP it doesn't play on the phone. I only hear some sound bytes that's all. I have checked with wireshark the media streamed is G.729 with RTP payload=110 bytes(=90 seconds framesize). I have also used various framesize for example 20,30,60 for testing, Modified "Preferred G.729 Millisecond Packet Size" parameters in Call Manger Service Parameter without any positive results. And I have verified the audio stream from another JMF/VoiceAge rtp client and I can hear the media perfectly fine.

When I use G.723/GSM codec with mode=send I get the following error message on the ip phone log.
313: WRN 04:28:15.539364 JVM: Startup Module Loader|Start Media Handler:? - Start Media  Error: type: UnavailableResource Create Send Stream Failed  Http Error code 400

send XML:
<startMedia><mediaStream><type>audio</type><codec>G.729</codec><mode>send</mode><address>225.3.4.5</address><port>25000</port></mediaStream></startMedia>

receive XML:
<startMedia><mediaStream receiveVolume="100"><type>audio</type><codec>G.729</codec><mode>receive</mode><address>225.3.4.5</address><port>25000</port></mediaStream></startMedia>

Phone Model:7971G
Firmware version: SCCP70.9-1-1SR1S

CCM Version:7.1.5.32900-2

Has anyone worked with RTP Streaming API with VoiceAge or any other G.729 codec provider?

Appreciate any help on this matter.

Thanks,
Samit

You can invoke RTP streaming via URIs in services. You can instruct the phone to transmit or receive an RTP stream with the following specifications:
•RTPRx
•RTPTx
•RTPMRx
•RTPMTx

NoteFor some Cisco Unified IP Phone models, the RTP Streaming URIs have been deprecated by the RTP Streaming API. See the “RTP Streaming API” section on page 4-1.

The supported format of the RTP stream is as follows:
•The codec is G.711 mu-Law.
•The packet size is 20 ms.
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This is a text from

Cisco Unified IP Phone Services Application Development Notes
Supporting XML Applications
Release 7.1(3)

It's a JMF problem. 
JMF sends RTP packet without to put a delay between packets.  G.729 requires 20 ms delay between packets.
Between 2 IPPhone one in receive mode and the other in Transmit mode, I can see with Wiresharl the correct G.729 payload.