CODEWITHSUNDEEP
asterisk
snirali
snirali

Reputation: 139

peer to peer RTP call setup in asterisk

I want to set up call between to peers in asterisk in which RTP flow is between two peers when internal calls.I don't want to go RTP flow from peer-asterisk-peer.I want to setup RTP flow like peer-peer.My sip.conf setup is like this.`

    [1001]
    type=friend
    host=dynamic
    nat=no
    ; qualify=no
    dtmf=rfc2833
    secret=1234
    callerid=1001 <1001>
    directrtpsetup=yes
    context=test_rtp
    disallow=all
    allow=g729
    allow=alaw
    allow=ulaw

Second peer 1002 is same.In this case RTP goes through asterisk.What changes should made to get peer to peer RTP call setup?

Upvotes: 0

Views: 3616

Answers (1)

MichelV69
MichelV69

Reputation: 1212

You're part of the way there already with the

directrtpsetup=yes

configuration option.

For this to work, you'll also need to add the following line:

directmedia=yes

You also have to ensure you do not use any of the following Dial() command options:

"t", "T", "h", "H", "w", "W" or "L"

... since Asterisk must remain in the audio path to handle these.

Lastly, both ends of the call must support and be configured for directmedia. If one end is not or does not, then Asterisk will remain in the audio path.

further reading

(if this answer addresses your question, please 'accept' it so that other users can benefit from it. thanks!)

Upvotes: 1

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