回復 6847# hersvim


According to your explanation and if we want to achieve the direct exchange of RTP voice packets beetween two end users, shall we need to enable the paramenter of canreinvite=yes instead of canreinvite=no suggested by some Asterisk variants out there like Trixbox and Elastix?

As far as I know, by default, canreinvite in Asterisk will route the media streams from SIP endpoints through itself.  Enabling this option causes asterisk to attempt to negotiate the endpoints to route the media streams directly, bypassing asterisk.  However, it is not always possible for asterisk to negotiate endpoint-to-endpoint media routing.

Thanks

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回復 6851# hersvim


   
Thanks for elaborating the merit and demerit of using CANREIVITE to us.  I have read the passage in 2004 that you advised.  In the last paragraph of the post, I also found that there is still a limitation of using "canreinvite=yes" if our devices are set behind NAT which will cause one audio stream when they communicate to the one on the outside.  In fact, many of the members here had experienced this before.  Can we expect a better suggestion to resolve the problem when we deploy our Asterisk Server and other related ATA or IP phones behind NAT?  

BTW, I've just learnt from http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite that the parameter of "canreinvite=yes" has a change in certain extent in the migration from Asterisk 1.2 to 1.4.

QUOTE:

The "canreinvite" option has changed. canreinvite=yes used to disable re-invites if you had NAT=yes. In 1.4, you need to set canreinvite=nonat to disable re-invites when NAT=yes. This is propably what you want. The settings are now: "yes", "no", "nonat", "update".

UNQUOTE


Thanks for sharing.

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其實我都想知道是什麼原因

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