Hi Guys,
Currently, i am facing problem with direct media/rtp. I installed Elastix 2.4 and Polycom Soundpoint IP Phones. I am trying to by pass the rtp/media not thru Elastix server. But it works only one way, the voice/rtp only flow from one IP Phone to other IP Phone. I read many articles about this rtp on Asterisk which stated it works. But why i can not. The parameters directrtpsetup=yes, recaninvite=yes already done.
Anyone can help me to make it works for me?
Thanks.
Sinjo
I am not sure if directrtpsetup is considered reliable. It certainly wasn’t at one. However it should not be necessary.
canreinvite should be directmedia, although I don’t think that recognition of the old name has yet been removed, and, in any case, the fact that you get one way audio suggests you are getting some direct media.
I would remove directrtpsetup and change the name of canreinvite. If you still have problems, the normal culprits are firewalls and NAT, but a copy of the SDP exchange from sip set debug on output is likely to be needed.
Hi David,
Thank youd for your help.
I already change to canreinvite=yes. But still one way audio, the audio always come only from the initiate caller. The called party audio can not transmitted.
I said:
Delete directrtpsetup
Delete canreinvite
Add directmedia=yes (the new name for canreinvite).
I also said if that still fails, provide copies of the SDP. Nearly every SIP problem here ends up with requests for the SDP, so you should be able to find many examples of how to get it.