SSRC =0x0 In RTP

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Asterisk Users 3 Comments

Hello, I have a problem where on an outgoing call a Grandstream phone (GXP2130)
closes the incoming voice stream about 1 second into the call (the remote party hears the Grandstream, the Grandstream doesn’t hear thr remote party). I have verified with logs and traces that this is not a NAT issue or any other network-related problem. All incoming RTP packets arrive at the phone on the correct port etc. as declared in the SDP. I opened a ticket with Grandstream and they replied: “

*the phone starts receiving RTP with SSRC =0x0 which is wrong”.*

Is this an Asterisk problem or the phones? Is this something that can be fixed on the Asterisk side?

Thank you,

Harel

3 thoughts on - SSRC =0x0 In RTP

  • Asterisk would be sending the RTP to the Grandstream. I’d suggest getting a packet capture using tcpdump or wireshark to confirm what they’ve said though. I just looked at the code and I don’t see a way that we’d ever have the SSRC be 0.

    Cheers,

  • in text format (server IP has been changed to 111.111.111.111). Some repeating RTP packets has been truncated. You can see that after the 200 OK SSRC is sent from the server to the phone as ‘0x0’. The same has happened with G729 codec.

    the SDP.

    Asterisk would be sending the RTP to the Grandstream. I’d suggest getting a packet capture using tcpdump or wireshark to confirm what they’ve said though. I just looked at the code and I don’t see a way that we’d ever have the SSRC be 0.

    Cheers,

  • I’m sorry but this version is old enough that what I currently know is far past it. It may have been possible in that old version for the SSRC
    to be as you state. In recent stuff it doesn’t seem to be possible.

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