I have done some further research, the first RTP packet is sent when playback() is called. No others. The application is running, if I press a key and goto a different item that would cause a new playback()/background() 1 more RTP packet is sent. To be clear If I call myself, RTP packets are sent. During a wait no packets are sent, only when playback() starts, and then only the 1 packet. This is true of any sip phone I have tried, whether or not it is local or remote. I can also call out through asterisk and that works, it just appears to be having a problem sending packets if it has to create the noise. As such this makes asterisk less than usable in my given situation. -- Trixter http://www.0xdecafbad.com UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 881 8487 FreeWorldDialup: 635378 -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050414/02125536/attachment.pgp
trixter http://www.0xdecafbad.com wrote:> I have done some further research, the first RTP packet is sent when > playback() is called. No others. The application is running, if I > press a key and goto a different item that would cause a new > playback()/background() 1 more RTP packet is sent. > > To be clear If I call myself, RTP packets are sent. During a wait no > packets are sent, only when playback() starts, and then only the 1 > packet. > > This is true of any sip phone I have tried, whether or not it is local > or remote. I can also call out through asterisk and that works, it just > appears to be having a problem sending packets if it has to create the > noise. As such this makes asterisk less than usable in my given > situation.Is Asterisk getting a stream of RTP packets from the SIP client? What happens if you start talking on the SIP device? Does Asterisk then start sending RTP? It still sounds like VAD and silence supression is enabled on the SIP device. -- Always do right. This will gratify some people and astonish the rest. Mark Twain
On Thu, 2005-04-14 at 07:23 -0500, Eric Wieling wrote:> Is Asterisk getting a stream of RTP packets from the SIP client? What > happens if you start talking on the SIP device? Does Asterisk then > start sending RTP? It still sounds like VAD and silence supression is > enabled on the SIP device. >Let me recap since it was spread over a couple emails back and forth between someone and this is a high volume list. Any sip clinet (local or remote) connecting to my asterisk system is met with silence. All sip clients send a RTP stream correct Asterisk reads DTMF from sip clients with no problem and executes the appropriate dialplan entries asterisk will hang on a playback() or background() call and will send only 1 RTP packet to the SIP slient (note the sip client is sending RTP the whole time, thus I do not think its silence detection on the client) If I press a number that causes a different playback() asterisk will send 1 more RTP packet and hang on that playback. If I go through asterisk but the destination is a sip client the RTP stream works perfectly. The *only* time there is a RTP problem is when the destination is asterisk. musiconhold sends no rtp packets, playback 1 packet only. SIP traffic goes back and forth, a sip show debug reveals proper call setup/tear down. It does not matter if the sip client is local or remote. This setup was working a couple days ago, which makes me curious as to what could possibly be happening. -- Trixter http://www.0xdecafbad.com UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 881 8487 FreeWorldDialup: 635378 -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050414/7768b24c/attachment.pgp