Interconnecting two Elastix servers, codec?

joshua

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#1
Hello,
I have setup two test Elastix servers and would like to setup an "Interconnect" as specified in Elastix without Tears, chapter 30.2.

My question is, how can I specify the link to use a specific codec, ie G729a?

Cheers
 

Bob

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#2
Joshua,

Under the trunks that you have setup for the interconnection under the PEER details on both sides of the interconnection make sure the following lines are in place

disallow=all
allow=g729


The only issue is that you must have the G729 codec installed as it does not come standard (it is a licenced product and purchased for each channel that you are likely to use).

If you do not have the G729 codec, can I suggest that you try with the GSM codec using the following settings

disallow=all
allow=gsm


Regards

Bob
 

rahul.rajan

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#3
Hi,

Is G.711 free to use or i need to purchase a license too.

By default which is the codec (if required) used by asterisk (elastix)

Could i also knw what is the RTP packet size when a call is established. (SIP to SIP call)

Kind Regards,
Royd Mendes
 

dicko

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#4
rahul.rajan:
greetings

g711 is free (but I'm sure AT&T or its descendants will take your money if you offer it), it is the de facto codec used for telephony for 50 odd years (no compression, just pre/de emphasis which is slightly different depending on the g711 variant)

There is no default codec, but one is normally required, (however theoretically I believe you could negotiate SLIN, but I guess even that is technically a codec),it is mutually negotiated between the two endpoints at the beginning of the session (SIP=Session Initiation Protocol). You can set your codec preferences in your sip.conf files (and its inclusions), where the last inclusion/exclusion/preference is presented first. The other endpoint can either honor or deny your requests, a "back-down" process is then negotiated until something mutually acceptable is agreed on. If no agreement can be made,both to codec and underlying protocol, the session will end.
The RTP packet size is commonly 1376 but SIP or the route between the endpoints can change that.
The codec rides within the RTP stream so negotiated. The underlying sip session, be it tcp or upd, (asterisk until 1.6 only speaks udp) polices and hopefully controls it.
 

rahul.rajan

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#5
Hey Dicko !

Thank for that details explantion.

The packet size 1376 is that in bytes ?

Thanks,
Royd
 

dicko

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#6
yes, so it will fit with all its baggage within ppoe or regular ethernet, I don't recommend its use for many other reasons within more restrictive transports, (but that is more a network issue than a sip issue apart from tuning the so called "jitter buffer" to suit ).
 

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