Dear Sir
Does the SIP server have any thing to do in order to select the CODEC used in the RTP session, or Its only rout the calls transperentlly.?
Thank you
Amr
CODEC Priority
Moderator: Brekeke Support Team
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voipwell.com
- Posts: 528
- Joined: Tue Sep 20, 2005 9:10 am
- Location: Tannersville, Pennsylvania
AMR,
The default behavior is that the sip server lets the two connection parties negotiate their own codec, but you can use the dial plan to intervene and force your chosen codec. This is what is so poweful about Brekeke.
The ability to force a codec and turn on rtp relay dynamically with the dial plan enables you to virtually make any connection between devices you need. For example Version 1 sip server can't relay T.38 packets so you can turn off rtp relay for faxes. Some phones can't negotiate codecs properly so you can force all incoming call to a particular phone to come in with g729. If you want all outgoing calls from a phone to use g729 and can't rely on the phone or the person managing the phone to set it up correctly you can force all outgoing calls from a particular phone to use g729.
The more you learn about Brekeke the more you realize the capabilities they give you are priceless.
The default behavior is that the sip server lets the two connection parties negotiate their own codec, but you can use the dial plan to intervene and force your chosen codec. This is what is so poweful about Brekeke.
The ability to force a codec and turn on rtp relay dynamically with the dial plan enables you to virtually make any connection between devices you need. For example Version 1 sip server can't relay T.38 packets so you can turn off rtp relay for faxes. Some phones can't negotiate codecs properly so you can force all incoming call to a particular phone to come in with g729. If you want all outgoing calls from a phone to use g729 and can't rely on the phone or the person managing the phone to set it up correctly you can force all outgoing calls from a particular phone to use g729.
The more you learn about Brekeke the more you realize the capabilities they give you are priceless.
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amr_elsawi
- Posts: 54
- Joined: Sun Nov 19, 2006 3:50 am
- Location: Kuwait
-
voipwell.com
- Posts: 528
- Joined: Tue Sep 20, 2005 9:10 am
- Location: Tannersville, Pennsylvania
If you want to force all calls to g729 then put this string in the deploy of your to and from pbx dial plans.
&net.rtp.audio.payloadtype=18
If you only want it for selected users or isp's then read the dial plan on how to change the dial plans.
http://www.brekeke-sip.com/download/bss ... lan_en.pdf
&net.rtp.audio.payloadtype=18
If you only want it for selected users or isp's then read the dial plan on how to change the dial plans.
http://www.brekeke-sip.com/download/bss ... lan_en.pdf