Rtp dynamic payload type 123

Hello,
We have a problem about RTP dynamic payload type 123. If you know, please explain when cisco gateways are nagotiation between non-cisco media gateways. When we look at rtp packets, Wireshark tag this packets malformed. Is there a way Can we disable between gateways are nagotiation with dynamic payload type 123.
i found only below info about rtp dynamic payload type 123 that is used for indicating Cisco CAS information.
RTP Payload Value
RTP Payload Definition
123
Indicates Cisco CAS information.
regards,
Omer

What is your hardware?
Do you have already see this doc?
Enable RTP Signaling on AS5350 and AS5400
In order to prevent errors caused by RTP packets of payload type “123” on Cisco AS5350 and AS5400 series platforms, RTP signal processing is disabled by default. Under some circumstances, packets of this type can cause an invalid memory address error in AS5350 and AS5400 series platforms, potentially crashing the devices.
On these models, you can enable RTP signal processing using the voice-fastpath voice-rtp-signalling enable hidden configuration command. However, before you enable RTP signal processing, prepare the platform to handle RTP packets of payload type “123” by enabling T-CCS.
After you prepare the platform, you can use these commands in order to enable or disable RTP signal processing.
In order to enable RTP signal processing, use this command:
Router(config)#voice-fastpath voice-rtp-signalling enable
In order to disable RTP signal processing, use this command:Router(config)#no voice-fastpath voice-rtp-signalling enable

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Dynamic rtp payload type 123

Hello,
We have a problem about dynamic rtp payload type 123. Wireshark is shown rtp packets which dynamic rtp payload type is 123 malformed.
why do cisco gateways send dynamic rtp payload type 123 packets to non-cisco gateways? Can we disable dynamic rtp payload type 123 nagotiation?
please help,
Thanks
Omer

Hello again,
Signaling Protocol is H323. I can not take debug h323 etc logs because of the device has too many calls. But you can find below rtp debug that i took before.
I can not share AS full config because privacy stuff.
voice service voip
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw
h323
emptycapability
h225 timeout setup 10
modem passthrough nse codec g711ulaw
sip
bind control source-interface Loopback3
bind media source-interface Loopback3
no voice-fastpath enable
dial-peer voice 1 voip
description INCOMING Traffic
service incomingcallcontrol
voice-class aaa 3
voice-class codec 101
incoming called-number B00290..........$
dtmf-relay rtp-nte h245-signal h245-alphanumeric
fax-relay ecm disable
fax rate 14400
ip qos dscp cs3 signaling
no vad
RTP(54397): fs rx s=10.10.11.1(20928), d=10.10.10.1(23946), pt=123, ts=1303D47D, ssrc=20351C0, marker=0
RTP(54397): fs tx s=10.10.10.1(23690), d=172.20.60.2(12994), pt=123, ts=1303D47D, ssrc=20351C0, marker=0
RTP(54397): fs rx s=10.10.11.1(20928), d=10.10.10.1(23946), pt=123, ts=1303D47D, ssrc=20351C0, marker=0
RTP(54397): fs tx s=10.10.10.1(23690), d=172.20.60.2(12994), pt=123, ts=1303D47D, ssrc=20351C0, marker=0
RTP(54397): fs rx s=10.10.11.1(20928), d=10.10.10.1(23946), pt=123, ts=1303D47D, ssrc=20351C0, marker=0
RTP(54397): fs tx s=10.10.10.1(23690), d=172.20.60.2(12994), pt=123, ts=1303D47D, ssrc=20351C0, marker=0
RTP(54397): fs rx s=10.10.11.1(20928), d=10.10.10.1(23946), pt=123, ts=1303D47D, ssrc=20351C0, marker=0
RTP(54397): fs tx s=10.10.10.1(23690), d=172.20.60.2(12994), pt=123, ts=1303D47D, ssrc=20351C0, marker=0
RTP(54397): fs rx s=10.10.11.1(20928), d=10.10.10.1(23946), pt=123, ts=1303D47D, ssrc=20351C0, marker=0
RTP(54397): fs tx s=10.10.10.1(23690), d=172.20.60.2(12994), pt=123, ts=1303D47D, ssrc=20351C0, marker=0
RTP(54397): fs rx s=10.10.11.1(20928), d=10.10.10.1(23946), pt=123, ts=1303D47D, ssrc=20351C0, marker=0
RTP(54397): fs tx s=10.10.10.1(23690), d=172.20.60.2(12994), pt=123, ts=1303D47D, ssrc=20351C0, marker=0
RTP(54397): fs rx s=10.10.11.1(20928), d=10.10.10.1(23946), pt=123, ts=1303D47D, ssrc=20351C0, marker=0
RTP(54397): fs tx s=10.10.10.1(23690), d=172.20.60.2(12994), pt=123, ts=1303D47D, ssrc=20351C0, marker=0
RTP(54397): fs rx s=10.10.11.1(20928), d=10.10.10.1(23946), pt=123, ts=1303D47D, ssrc=20351C0, marker=0
RTP(54397): fs tx s=10.10.10.1(23690), d=172.20.60.2(12994), pt=123, ts=1303D47D, ssrc=20351C0, marker=0
RTP(54397): fs rx s=10.10.11.1(20928), d=10.10.10.1(23946), pt=123, ts=1303D47D, ssrc=20351C0, marker=0
RTP(54397): fs tx s=10.10.10.1(23690), d=172.20.60.2(12994), pt=123, ts=1303D47D, ssrc=20351C0, marker=0
RTP(54397): fs rx s=10.10.11.1(20928), d=10.10.10.1(23946), pt=123, ts=1303D47D, ssrc=20351C0, marker=0
RTP(54397): fs tx s=10.10.10.1(23690), d=172.20.60.2(12994), pt=123, ts=1303D47D, ssrc=20351C0, marker=0
RTP(54397): fs rx s=10.10.11.1(20928), d=10.10.10.1(23946), pt=123, ts=1303D47D, ssrc=20351C0, marker=0
RTP(54397): fs tx s=10.10.10.1(23690), d=172.20.60.2(12994), pt=123, ts=1303D47D, ssrc=20351C0, marker=0
RTP(54397): fs rx s=10.10.11.1(20928), d=10.10.10.1(23946), pt=123, ts=1303D47D, ssrc=20351C0, marker=0
RTP(54397): fs tx s=10.10.10.1(23690), d=172.20.60.2(12994), pt=123, ts=1303D47D, ssrc=20351C0, marker=0
RTP(54397): fs rx s=10.10.11.1(20928), d=10.10.10.1(23946), pt=123, ts=1303D47D, ssrc=20351C0, marker=0
RTP(54397): fs tx s=10.10.10.1(23690), d=172.20.60.2(12994), pt=123, ts=1303D47D, ssrc=20351C0, marker=0
RTP(54397): fs rx s=10.10.11.1(20928), d=10.10.10.1(23946), pt=123, ts=1303D47D, ssrc=20351C0, marker=0
RTP(54397): fs tx s=10.10.10.1(23690), d=172.20.60.2(12994), pt=123, ts=1303D47D, ssrc=20351C0, marker=0
RTP(54397): fs tx s=10.10.10.1(23690), d=172.20.60.2(12994), pt=123, ts=1303D47D, ssrc=20351C0, marker=0
RTP(54397): fs rx s=10.10.11.1(20928), d=10.10.10.1(23946), pt=123, ts=1303D47D, ssrc=20351C0, marker=0
Regards

Add dynamic Payload Type to JMF

Hi all
I am trying to write a Java application to play streaming video (RTP with dynamic payload of 96) from UCT IPTV server, http://uctimsclient.berlios.de/uctiptv_howto.html
(My application could receive the streaming RTP but cannot play it)
I think that I need to register 2 formats to JMF:
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ffenc_h263p Caps:
video/x-raw-yuv, format=(fourcc)I420, width=(int)352, height=(int)240, pixel-aspect-ratio=(fraction)8/9, framerate=(fraction)30000/1001
Video Caps:
application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)H263-1998, payload=(int)96, ssrc=(guint)2779507444, clock-base=(guint)599556584, seqnum-base=(guint)5482
Audio Caps:
application/x-rtp, media=(string)audio, clock-rate=(int)90000, encoding-name=(string)MPA, payload=(int)96, ssrc=(guint)3836987939, clock-base=(guint)2707891566, seqnum-base=(guint)52755
I don't know much about codecs. Could any one please help me out with this problem.
Thanks
gladiator_vn

Here is one of the coolest things about LV, and one of its major advantages over languages like VB - it has automatic memory management and it compiles whenever you make a change. This allows you to be very free in the way you handle data and change your code. In this case, you should note the small gray dot that appears when you wire the dynamic data into the Array of Waveforms input. This gray dot means that LV coerces the source data type (DDT, in this case) to the target data type (Array of waveforms). If it wouldn't match, you would get a broken wire, the VI couldn't run, and if you hover over the wire or press the broken run button you would get an exact description of the error. Cool, eh?
This can also be seen in other things. For example, you can change an array's data type or number of dimensions by simply doing it. This will automatically be reflected anywhere the array is called in the program, and if anything won't match, the VI will be broken. If you change the # of dimensions of an array, any array function which that array is wired to will adapt. This allows you to be very safe in the way you write code, because you have LV watching all that annoying stuff for you. You don't have to worry about misspelling a variable here (for the most part).
One important point about coercion - it does take processor power. This will usually only be evident in large arrays or loops, but it is an important point. Another thing is that coercing numeric types can cause you to lose data - coercing a DBL to a U16, for example, will cause you to lose all the fractional data.
To learn more about LabVIEW, I suggest you try searching this site and google for LabVIEW tutorials. Here and here are a couple you can start with. You can also contact your local NI office and join one of their courses.
In addition, I suggest you read the LabVIEW style guide and the LabVIEW user manual (Help>>Search the LabVIEW Bookshelf).
Try to take over the world!

Firefox webrtc rtp payload type

I'm seeing an issue when placing a webrtc call with firefox when the answer SDP has different payload types than the offer. for example, i get an offer from firefox that has the opus codec with payload type 109. if the answer SDP comes in with the opus codec and payload type 110, audio will not be heard. looking in the WebRTC.log file you will see a whole bunch of "IncomingRTPPacket received invalid payloadtype".
i believe the correct behavior in this case is for firefox to transmit opus with RTP payload type 110 (ie the remote payload type) and expect to receive opus with payload type 109. it looks like firefox will send with 110 and expect to receive with 110.
the same scenario with chrome does not have a problem. if i change the SDP answer to match the payload type of the offer, audio can be heard in firefox.
the same issue can be seen with VP8 video or other codecs with dynamic payload types.
for reference, the relevant excerpt from rfc3264:
"In the case of RTP, if a particular codec was referenced with a
specific payload type number in the offer, that same payload type
number SHOULD be used for that codec in the answer. "
in this case the answer SDP is violating the SHOULD in the rfc, but this is sometimes unavoidable when interworking SDP between devices (ie some sort of back-to-back signaling agent).

That RFC is a proposed standard at the moment [http://datatracker.ietf.org/doc/rfc3264/]
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RTPExport using a new payload type

I'm interested in using the RTPExport example:
http://java.sun.com/products/java-media/jmf/2.1.1/solutions/RTPExport.html
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Hello,
I am using the AVReceive2 program (http://java.sun.com/products/java-media/jmf/2.1.1/solutions/AVReceive.html#requirements) as posted to hopefully play back RTP in real time. I have it set to receive a known good RTP stream and I receive the following error:
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I have seen posts for a similar error but it seemed to be in the context of an A-law add on which am I not doing. Am I missing a plug-in or modification to the code?
thanks in advance,
Erich

Hi Erich,
I have the same problem when I using AVReceive2.java .Could yuo describe your solution in order to solve the issue?
In order to create the RTP stream I used VLC and is impossible to change the encapsulation method and then the payload type .
What kind of streaming server are yuo used?
My e-mail address is [email protected]
If you can , please write me your proceeding in order to solve the problem.
Thank you

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I am trying to use JMF to play back recorded RTP streams containing RTP payload type 8 (ALaw) and 13 (CN). It seems these are not supported by default.
Did anybody add support to JMF to support these payload types succesfully? Help is greatly appreciated.

Hello
i have a same problem.so i was asking if you solved it or not.
Thanks

Can someone help me with "no format has been reg. for RTP Payload type 10"

I am trying to send audio from Mac to PC. Mac uses quicktime for java and PC uses jmf. I looked at the Jmf supported formats list and on the SDP file for mac i used payload types like :
m=audio 2656 RTP/AVP 0
c=IN IP4 239.60.60.60
. I also tried other formats too but I always get "no format has been registered for rtp payload type 10". on the pc side. Anyone can help me out? thanks!

Sprint's unlimited wireless service is actually ten bucks more a month than my Verizon service, according to Sprint's Web site.
And Verizon's customer support is the best in the industry.
I have never experienced a "throttling" with my Verizon account. And I have never experienced any limited bandwidth usage problem.

Payload type of wav files encoded with gms codec

Hello everybody,
I have an application which receives and plays alaw formated wav files using JMF(with the help of some additional libraries) . I want to use lower bandwidth for my application. A quick google search indicated that i can do this by using gsm format.
I have 2 questions:
1-) what are the differences between .gsm files and .wav files which are encoded with gsm codecs? Which one's quality is better? which one's compression ratio is better?
2-) which payload type should i use for wav files which are encoded with gsm codecs?
In [Rtp Payload list|http://www.iana.org/assignments/rtp-parameters] page, there is a gsm entry whose payload is 3. I suppose that is for files with gsm extension. I tried it but it didn't work with my wav sample (as u can guess it was encoded with gsm 6.10 codec)
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Thanks in advance
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Edited by: Bugra.Hasbek on Nov 10, 2008 1:18 AM

Bugra.Hasbek wrote:
Hello everybody,
I have an application which receives and plays alaw formated wav files using JMF(with the help of some additional libraries) . I want to use lower bandwidth for my application. A quick google search indicated that i can do this by using gsm format.
I have 2 questions:
1-) what are the differences between .gsm files and .wav files which are encoded with gsm codecs? Which one's quality is better? which one's compression ratio is better?I would imagine they are the same...
2-) which payload type should i use for wav files which are encoded with gsm codecs?I would imagine 3...
In [Rtp Payload list|http://www.iana.org/assignments/rtp-parameters] page, there is a gsm entry whose payload is 3. I suppose that is for files with gsm extension. I tried it but it didn't work with my wav sample (as u can guess it was encoded with gsm 6.10 codec)
Any suggestion is appreciated. If you have a comment about the main problem (which is using lower bandwidth for sending audio data) i would also like to hear itMy suggestion would be not to worry about so much stuff. JMF does all of this for you automaticlly.
On the sender side, make your processor take in whatever kind of supported file you want. Set the processor to output your GSM mono, and then you can just use a Player to play it on the other side. It's really not very different than the examples...
[http://java.sun.com/javase/technologies/desktop/media/jmf/2.1.1/solutions/AVTransmit.html]
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[http://java.sun.com/javase/technologies/desktop/media/jmf/2.1.1/solutions/AVReceive.html] should be able to play this out of the box...

Accessing object payload type attributes by index?

Plz help
Using object type in queues enforce some limitations to altering object type. You can use only ALTER TYPE and use alteration that only "expand" properties. It'is possible to change column type from VARCHAR2(10) to VARCHAR2(20) but not vice versa. Also it's impossible to drop object attribute or re-create object type. While storing object source in svn it's uncomfortable to view object source as sequence of ALTER TYPE commands. So, it seems that solution can be usage XMLTYPE as object payload type. But this raise another problem.
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Quicktime Dynamic Payload Support

Hi -
I have created a streaming server using the RTSP/SDP/RTP protocols and I am having a little trouble getting a dynamic payload to work with my server. I am trying to stream a mu-law encoded file at 11025 Hz sampling rate and 8 bits per sample. The standard is 8000 Hz and my server handles that just fine. However, when I try and stream the 11025 Hz audio Quicktime tries to connect to the internet because it doesn't understand the file. When I save the file to my computer and play it Quicktime plays the file back just fine.
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I am not sure if the PCMU title is correct for this stream.
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Perian should add Flash support to QuickTime
<http://perian.org>

AVReceive2 Sample - Payload Type 13 (Comfort Noise)

I'm getting an error from an RTP Stream changing to payload type 13...
Is there any way around this problem, I want to record the stream...
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I need something lightweight...
Thanks,
Ted

Is there any way around this problem, I want to record the stream...Yeah, go in and implement some code to handle the LocalPayloadChangeEvent when it's received by the SendStreamListener...
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JMF Payload type 33

Hey guys, i m trying to play RTP stream video but everytime i play, it give me the following error.
No format has been registered for RTP Payload type 33
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Hi,
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Anyway, for more information on the payload type, please refer to RFC3551.
Best Regards
Ferdinand Ng

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Currently, I have the following basic VI setup to save Data from my PCI6221 Data Aquisition Card. The problem I'm having is I keep getting the last iteration of the while loop in the measurement file and that's pretty much it. When I try to index the data leaving the loop it gives me a 2D array of Data which cannot be input into the "Write to Measurement File" VI. How would I save this to a useful Data/time step format? I was wondering of a way to continuously collect the Data and then save it in a large measurement file that I would manipulate in Matlab/excel? Am I using the wrong type of loop for this application? I also noticed my Dynamic Data array consists of data, time, timestep and then a vector of the data taken. Is it possible to just get a vector of the time change per sample alongside the data? Sorry for the barrage of questions but any help would be greatly appreciated, and thanks in advance!
-Bryan
Attachments:
basic DAQ.vi ‏120 KB

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Darren Nattinger, CLA
LabVIEW Artisan and Nugget Penman

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chapters 12 - 16 have quite a lot of examples on JMS.
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null

Rtp dynamic payload type 123

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