"ast_rtp_write: No remote address on RTP instance.... so dropping frame"

General Help
1

Hi Team,
I noticed the above message during a “core set debug 1”

Could you help me to address the problem?

Regards

3

Hi david555, I try to answer to each questions:

Are you actually experiencing a problem with calls? If so what? If you are referring to voice problems, the answer is no

Are you using SIP or WebRTC? We have people that use in LAN a Grandstream phone. People at home use CiscoAny Connect for VPN and the client sip is MicroSIP

Are you using chan_sip or chan_pjsip? chan_sip

In which direction was the call set up? It is a call centre system that receives Inbound calls that are assigned to SIP/xxx extension logged on queue

If inbound, is it early or late offer SDP? I don’t know how verify this point

Is this early media or normal media? I don’t know how verify this point

Sorry for me English, I’m not native

Thanks,
Angelo

5

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