Disabled sent rtp packet

Hi Guys,
Anyone can help me to disabled sent rtp packets from asterisk 11?? i use canreinvite=no or directmedia=nonat but it didnt work.

Asterisk sometimes needs to send RTP!

For direct media to be possible, when Asterisk is not the original source of the RTP:

directmedia needs to be enabled on both parties. For nonat, both parties must be on the local network.

They must have some codecs in in common and those codecs must be allowed by sip.conf.

Neither party may have any feature enabled that would require Asterisk to look for DTMF digits during the call, or to record the call.

The two SIP channels must be bridged directly to each other.

Generally it is better to provide specific configurations and logs, rather than asking open ended questions about what could be wrong.

Thanks David
I create a room by meetme as if only one persone can talk in room at any moment my problem is that rtp pakets by type 00 sent frequently by asterisk when nobody talking in room and this used Bandwidth. Is it possible asterisk send rtp only in media transfer or when get rtp packets?

You are talking about silence suppression, not direct media. I’ll leave Malcolm to tell you whether confbridge supports this, or you could look at the sample configuration file. I’m pretty sure that meetme doesn’t, and Asterisk generally doesn’t like silence suppression.

According to you I used confbridge instead meetme but asterisk frequently sent rtp packet yet!!! i uncomment dsp_drop_silence=yes in confbridge.conf but nothing changed :frowning: can i increase the time of sent this packets by asterisk??

Asterisk will always send a constant stream of packets. The only way to adjust the time between packets is to adjust the packetization rate (append a colon and then the rate to the codec you’re using for the peer in sip.conf, note that not all rates are supported for all codecs).

Thanks a lot Malcolmd
I use ulaw:150 and alaw:150 in general section in sip.conf file but its max rate is 150ms!!! have you another suggested for reduce bandwidth consumption in silence mode??

Nope. Asterisk will always send a constant stream of packets.

In your opinion can i solved my problem by change meetme source file??