[asterisk-dev] a=sendrecv in 183 Session Progress

Abelenda Diego diego.abelenda at domo-safety.com
Mon Sep 3 08:10:03 CDT 2012


On Mon, 3 Sep 2012 05:37:29 +0200
Pavel Troller <patrol at sinus.cz> wrote:

> Hello,
> 
> > Hello,
> > I'm wrong or it should be a=sendonly in 183 Session Progress ?
> 
> IMHO it's better to generally use sendrecv in the Early Media state,
> because it allows to transfer for example RFC2833 DTMF samples, which
> can be used for one of the implementations of the overlap dialling
> (not RFC 3578 based).
> 
> With regards,
>   Pavel
> 
> > 
> > Calling an application Playback with flag 'noanswer' generates 183
> > with a=sendrecv.
> > 
> > Here is a sip debug of 183:
> > <--- Transmitting (no NAT) to 192.168.245.25:5060 --->
> > SIP/2.0 183 Session Progress
> > Via: SIP/2.0/UDP 192.168.245.25:5060
> > ;branch=z9hG4bK7da3300e;received=192.168.245.25;rport=5060
> > From: "Anonymous" <sip:Anonymous at anonymous.invalid>;tag=as2fe7eeee
> > To: <sip:user01 at sip.testserver.lan>;tag=as7993bb65
> > Call-ID: 0616052b6bb4b4a45897ad1e706cc98f at 192.168.245.25
> > CSeq: 102 INVITE
> > Server: Asterisk PBX 1.8.11-cert7
> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
> > INFO, PUBLISH
> > Supported: replaces, timer
> > Contact: <sip:user01 at 192.168.245.10:5060>
> > Content-Type: application/sdp
> > Content-Length: 290
> > 
> > v=0
> > o=root 1550475731 1550475731 IN IP4 192.168.245.10
> > s=Asterisk PBX 1.8.11-cert7
> > c=IN IP4 192.168.245.10
> > t=0 0
> > m=audio 13218 RTP/AVP 0 8 101
> > a=rtpmap:0 PCMU/8000
> > a=rtpmap:8 PCMA/8000
> > a=rtpmap:101 telephone-event/8000
> > a=fmtp:101 0-16
> > a=silenceSupp:off - - - -
> > a=ptime:20
> > a=sendrecv
> 
> 
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Hello,

sadly as I experienced myself some days ago, there are some SIP to PSTN
bridges that absolutely need to receive audio before being able to
send the audio correctly. So the a=sendreceive is not an option here as
audio must be transmitted.
In my case the offending party was either an Asterisk server or behind
an Asterisk server.

See here for my emails:
http://lists.digium.com/pipermail/asterisk-dev/2012-August/056582.html
http://lists.digium.com/pipermail/asterisk-dev/2012-August/056602.html

Best regards,
Diego Abelenda




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