[OpenSER-Users] Changing Audio Ports

Daniel-Constantin Mierla daniel at voice-system.ro
Mon Aug 20 18:29:34 CEST 2007


Hello,

the sip device is broken, should update the destination port and IP from 
the 200 ok reply. The best is to ask the vendor to fix it. Think about 
para;lel forking, there could be many 183 coming from different 
addresses and ports, what would do the device?

Temporary solution is to drop any 183 or to use perl substitutions for 
such replies (all to be done in onreply_route).

Cheers,
Daniel

On 08/18/07 07:24, kelvin-lists at williamschadwell.com wrote:
> In a previous query I asked if someone could shed some light on as to why
> my endpoints do not receive any audio when the call is redirected to an
> announcement server.
>
> After a lot of testing, I believe I have found the problem.
>
> When I initiate a call from my end point the end point advises the callee
> as to the port the RTP traffic will be present.  When the call is handed
> to my PSTN gateway the Gateway responds with its port for RTP in a 183
> Session Progress.
>
> When that call fails (due to timeout) we want to send it over to Asterisk
> where an announcement will be played to the caller--however the caller
> never hears it.  The traces show that Asterisk advertises its RTP on a
> different port that that of the PSTN Gateway.  Some of my endpoints (Cisco
> IP Phone 7940 and Sipura devices) see and listen for the audio on the new
> advertised port, however my Arris EMTAs do not, it appears as though they
> are still "tuned in" to the original audio port advised by the PSTN
> gateway.
>
> My question, is is possible to strip away the "m=audio 22040 RTP/AVP 0 8
> 18 101." from the SIP message?  I would like to strip it away in the event
> of a 183 from my gateway (that advertises the port), but pass it when the
> call is actually answered.
>
> If it is not possible to strip the RTP port information away from the
> message, what would be the best way in handling a situation like this.
>
> Many thanks in advance.
> kw
>
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