sip video direct rtp

Discussion in 'General' started by tadaxi.chen, Jul 8, 2009.

  1. tadaxi.chen

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    two sip video phone(hard phone,realshow,product by my company http://www.putian-nst.com) can talk through elastix,but all rtp data(voice and video) exchange within elastix server.
    can this video call exchange rtp data each other,means p2p.
    I had tried these steps:
    1 in sip_genneal_custom.conf
    canreinvite=yes
    videosupport=yes
    allow=h264
    directsetuprtp=yes
    nat=yes

    2 canreinvite=yes nat=yes in extension files

    still rtp send and from elastix.
     
  2. dicko

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    tadaxi.chen:

    Elastix is based on Asterisk, so your question can only get a non-authoritative answer here (we wear Elastix hats here comfortably, for an Asterisk specific question I suggest the Digium forum is closer to the "source" ).

    However, with your settings, Your hardware's request should be honored by the underling asterisk process.
    If the endpoints are in the same network, nat=no is probably more appropriate
    If not in the same network then suspect your router/NAT setup.


    Otherwise . . .

    sip debug should allow your engineers to debug (as I'm sure you were aware) why your reinvites are not being honored, and by whom.

    regards, and please keep us informed of your progress with this hardware.

    dicko
     

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