Re: [Discuss-gnuradio] RAW source

[Sara Chérif]

Thanks Activecat very much .

Now, SIP packets length is varying & RTP packets have fixed length of 87 bytes . Now to input this traffic coming from the soft phone to the Gnuradio to be processed, i use the UDP source block(to read the UDP packets from the soft phone port)
  in UDP source block , I can determine only one port number and Ip address. I put the IP address of the Wlan interface (my laptop IP address) , i also tried to put "0.0.0.0" and the port number of twinkle soft phone on which i receive the RTP traffic.

I faced some problems:

First ,How to receive now the SIP signaling packets and the Rtp packets at the same time to establish a call with another soft phone ( as the soft phone uses 2 different port numbers for sip & RTP while i have to identify only one port number in the UDP source block)??
The flow graph i use consists of :Udp source - throttle -Uchar to float and WX GUI sink scope.
I tried to duplicate the flow graph (to use another one which is the same as above and runs in parallel )
Also I tried to use 2 UDP source block then the add block.
But I don't know till now how to multiplexes the two traffics (SIP and RTP) as we need to integrate the whole system to see this & what is the right method ? DO I have to write simply the port of the ethernet connection ?
What will I do in this case  : If I want to make the call between the 1st & 2nd lap ( One has Twinkle & the other has GNU Radio & Twinkle,  and I want GNU Radio to capture packets coming to Twinkle ? (use 2 udp source + add block ) ?

Second, i don't know how to determine the payload size. what if the packet length in the case of the Sip signaling is variant. In the  soft phone, we can determine the maximum transfer unit (MTU), does this help here ?

Third, i have a question relating to the OFDM transmitter: what is the required packet length , I wrote it 100  as each sip packet is 87 bytes and I will use coding with rate 7/8 then after coding each packet is 100 bytes) what if the packet length is variable as in the case i stated above (SIP pkt length is not constant)?

Note: i want to capture the packets coming from the soft phone in order to process them on gnuradio using OFDM system and send and receive them at the other end using two Usrps to be used as input to the other soft phone.

Thanks in advance

2014-06-24 8:05 GMT+03:00 Activecat <address@hidden>:

On Thu, Jun 19, 2014 at 9:32 PM, Sara Chérif <address@hidden> wrote:

Sorry , I forgot to say that some packets coming from Twinkle are SIP & ARP packets not only RTP.

Hence , I think I need to receive RAW packets in GNU Radio ( as I have different type of packets:  RTP , ARP , SIP packets)

Note that :

I use 4 laps & 2 USRPs.

1st & 4th lap has twinkle softphone & I will make a call between them.

2nd lap has an OFDM TX implemented by Gnuradio.

3rd lap has an OFDM RX implemented by Gnuradio.

First and 2nd laps are connected by Ethernet cable.

3rd & 4th laps are connected by Ethernet cable.

One USRP is connected to the 2nd lap , the other USRP is connected to the 3rd lap.
2nd lap(GNUradio ,ofdm tx)  will receive the packets from 1st lap(Twinkle) using RAW socket (as I think ) .

2nd lap will send the packets to the 3rd lap by the USRPs.

3rd lap ( Gnuradio , ofdm rx)  will send real time voice packets to 4th lap (Twinkle) using RAW socket(as I think).

If I want only to make a call & send and receive the voip real time packets , Do I have then to write a specific block as this mentioned decoder ?

Thanks in Advance.

Hi Sara,

After few rounds of communications, apparently you are interested to intercept the VoIP session at OSI layer 5 or above (eg. RTP session), and you try to use gnuradio (with USRPs) to accomplish that.

Probably this is because your area of interest is actually those RTP and VoIP stuffs.

I have few highlights:

1).  To accomplish this, one of the right ways is to make your PC becomes one of the VoIP endpoints. 

In this case you listen to the speaker and speak to microphone attached to your PC.  At the other end of the session is a SIP phone.

With this, you really dig into the RTP and VoIP stuffs.

You need neither gnuradio nor USRP.  In this case the wireless link is trivial in your setup.

2).  In this case you need ASIO [1] or something similar, and start working from the low level.

But as ASIO may not sound as an easy topic, you are looking for tools like gnuradio to abstract the low level stuff.

Yes gnuradio has built-in blocks that abstract ASIO, but that abstraction doesn't mean to let you work at RTP layer.

3).  In this case the gnuradio is not the right tool for your purpose.  You had chosen the tool without first qualifying it.

Your setup above (with the 4 laps) couldn't actually force gnuradio to intercept the VoIP session at RTP layer.

This is a wrong design.

Let's be honest about your area of interest.

Anyway I could be wrong.

1. http://www.boost.org/doc/libs/1_55_0/doc/html/boost_asio.html