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

[Pavel Troller]

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