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 To :  Yate mailing list <yate@v...>
 From :  Alfred Stainer <alfred.stainer@g...>
 Subject :  No RINGBACK
 Date :  Thu, 24 Jun 2010 18:14:08 +0200
Hi,
I have the problem of "no ringback".

The scenario is:

1. A sip phone (Grandstream BT201)
2. Yate (latest from svn)
3. Public ISDN network via Sangoma A108
4. Called phone

My SIP Phone sends the INVITE to YATE that reply with a 100 trying.
Yate sends a SETUP message to PSTN via ISDN E1 connection.
PSTN sends a CALL PROCEEDING
PSTN sends an ALERT message
YATE sends a 180 Alerting with SDP to caller Phone

In this situation I don't ear the ring tone because the caller phone has
received the SDP in 180 ringing but the ISDN audio path is not yet closed.

But why Yate sends the SDP message? The SDP message shouldn't be related to
ISDN "media"?

In the ISDN side the media is present after CONNECT message or when CALL
PROCEEDING, ALERT or PROGESS is received with an appopriate (ie.
in-band-info) progress description information element.

Another question: the q931.cpp file in the "processMsgCallProceeding" method
and in the "processMsgAlerting" method seems to ignore the information
element Progress Description that can contains essential info (i.e.
in-band-info). I'm wrong?

Thanks,

Alfred Stainer

Captured packets:



ISDN Alert:

Protocol discriminator: Q.931
    Call reference value length: 2
    Call reference flag: Message sent to originating side
    Call reference value: 0012
    Message type: ALERTING (0x01)



SIP 180 Ringing:

Session Initiation Protocol
    Status-Line: SIP/2.0 180 Ringing
    Message Header
    Message Body
        Session Description Protocol
            Session Description Protocol Version (v): 0
            Owner/Creator, Session Id (o): yate 1277387718 1277387718 IN IP4
-----------
            Session Name (s): SIP Call
            Connection Information (c): IN IP4 ---------
            Time Description, active time (t): 0 0
            Media Description, name and address (m): audio 27108 RTP/AVP 18
101
            Media Attribute (a): rtpmap:18 G729/8000
            Media Attribute (a): fmtp:18 annexb=no
            Media Attribute (a): rtpmap:101 telephone-event/8000



Hi,
I have the problem of "no ringback".

The scenario is:

1. A sip phone (Grandstream BT201)
2. Yate (latest from svn)
3. Public ISDN network via Sangoma A108
4. Called phone

My SIP Phone sends the INVITE to YATE that reply with a 100 trying.
Yate sends a SETUP message to PSTN via ISDN E1 connection.
PSTN sends a CALL PROCEEDING
PSTN sends an ALERT message
YATE sends a 180 Alerting with SDP to caller Phone

In this situation I don't ear the ring tone because the caller phone has received the SDP in 180 ringing but the ISDN audio path is not yet closed.

But why Yate sends the SDP message? The SDP message shouldn't be related to ISDN "media"?

In the ISDN side the media is present after CONNECT message or when CALL PROCEEDING, ALERT or PROGESS is received with an appopriate (ie. in-band-info) progress description information element.

Another question: the q931.cpp file in the "processMsgCallProceeding" method and in the "processMsgAlerting" method seems to ignore the information element Progress Description that can contains essential info (i.e. in-band-info). I'm wrong?

Thanks,

Alfred Stainer

Captured packets:



ISDN Alert:

Protocol discriminator: Q.931
    Call reference value length: 2
    Call reference flag: Message sent to originating side
    Call reference value: 0012
    Message type: ALERTING (0x01)



SIP 180 Ringing:
 
Session Initiation Protocol
    Status-Line: SIP/2.0 180 Ringing
    Message Header
    Message Body
        Session Description Protocol
            Session Description Protocol Version (v): 0
            Owner/Creator, Session Id (o): yate 1277387718 1277387718 IN IP4 -----------
            Session Name (s): SIP Call
            Connection Information (c): IN IP4 ---------
            Time Description, active time (t): 0 0
            Media Description, name and address (m): audio 27108 RTP/AVP 18 101
            Media Attribute (a): rtpmap:18 G729/8000
            Media Attribute (a): fmtp:18 annexb=no
            Media Attribute (a): rtpmap:101 telephone-event/8000