[asterisk-dev] Codec negotiation on wiki

[Sylvain Boily]

Hello,

I think the page (1) about codec negotiation in the wiki has an error 
with the dialplan function documentation. Looks like this function 
doesn't exist, i checked on source code and there is not dialplan 
function with this name.


This is probably PJSIP_MEDIA_OFFER and PJSIP_SEND_SESSION_REFRESH who 
are involved to play with the codec negotiation and dialplan or i 
probably missed something.


Sylvain

(1) https://wiki.asterisk.org/wiki/display/AST/Codec+Negotiation 

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Re: [asterisk-dev] Pjsip aor with multiple contacts

[Kevin Harwell]

On Mon, Apr 27, 2020 at 5:49 AM Learn  wrote:

> Hi guys,
>
> sorry just a simple question which i can’t find an answer online.. if i
> have multiple contacts listed in one aor section in pjsip.conf and have
> quality on which of the contacts will be used if both contacts are
> reachable?
>
> ie:
> [trunk]
> type = aor
> contact = sip:t...@domain1.com:5061
> contact = sip:t...@domain2.com:5061
> qualify-frequency = 30
>
> is it round-robin, random, in the order of the config file or the lowest
> RTT?
>

It's semi-random.

>From what I can tell it appears it will choose the "first" reachable
contact (includes dynamic and static contacts) it knows about. I put
"first" in quotes though because the contacts are put into an unsorted
container, and then the "first" reachable contact in the container is used.
So depending on how a particular contact is hashed, and stored in the
container affects its location in said container.

If all you have are static contacts then the same one should be chosen each
time. However when including dynamic contacts it's possible (once
registered, and reachable) it could become the one used.


>
> cheers,
>
> Roland
>
>
-- 
Kevin Harwell
Software Developer
Sangoma Technologies
Check us out at: https://sangoma.com & https://asterisk.org
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[asterisk-dev] Pjsip aor with multiple contacts

[Learn]

Hi guys,

sorry just a simple question which i can’t find an answer online.. if i have 
multiple contacts listed in one aor section in pjsip.conf and have quality on 
which of the contacts will be used if both contacts are reachable?

ie:
[trunk]
type = aor
contact = sip:t...@domain1.com:5061
contact = sip:t...@domain2.com:5061
qualify-frequency = 30

is it round-robin, random, in the order of the config file or the lowest RTT?

cheers,

Roland



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