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Extracting opus audio from rtp using wireshark

Tgorma200
Level 1
Level 1

I'm conducting a project where I need to send and receive simulated call audio between 2 servers. I'm going to be assessing the quality of these files using POLQA/PESQ and need to preserve their bandwidth at 48kHz. Unfortunately the PBX software Im using (Asterisk) cannot record audio higher than 8kHz, though it does support its transfer.

 

I had the idea to capture the audio from the stream directly instead using wireshark and I've found there are plenty of tools out there to help me. I'm struggling to find one that works. I've attempted to do something similar to this thread here, using opus-tools' opusrtc and opusdec functions to extract and decode the pcap files. 
When I do this I end up with unplayable or empty files though. Are there any other tools to accomplish this? 

Any advice would be greatly appreciated as I've been banging my head against this particular brick wall for a few days.

 

Tommy

2 Replies 2

Dennis Mink
VIP Alumni
VIP Alumni

although this is not really a cisco related post, do you actually see the call as a VOIP call in wireshark?

 

can you describe how you are going about capturing the RTP stream?

 

thanks

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@Dennis Mink wrote:

although this is not really a cisco related post, do you actually see the call as a VOIP call in wireshark?

 

can you describe how you are going about capturing the RTP stream?

 

thanks


Apologies for the lack of relation to cisco - I have tried a few other forums but as I see the best responses I found were on the thread I linked to. 

 

The VoIP calls show up without issue. I can view them in telephony -> voip calls and also in telephony -> RTP stream activity. I've tried exporting them as a pcap file and I've also been experimenting with the 48khz sip/rtp opus sample capture from wireshark. I'm aware they can also be exported as raw audio (.au) but these are limited to 8kHz and I need them at the original sample rate for my tests.

 

Im then running the pcaps of the rtp data through the opus-tools function using the " opusrtp --extract [filename]" command which produces a file (rtpdump.opus) I've tried decoding this using the opusdec function and it does succesfully output a file however this appears to be empty. I'm not sure if it may be due to packet header? Maybe I need to specify the start of the rtp payload but I'm not sure what the value for that should be exactly and have just been using the default setting.