Pjsip: 180 ringing SDP follow by 200/ok sdp

Asterisk Asterisk SIP
1

Hi all ,

I use Asterisk certified/13.18-cert3 with pjsip channel.

I have this scenario
Send INVITE with SDP (offer).
Server responds with 180 Ringing with SDP (answer). Here SDP points
to the Media Server that provides “Ring Back” tone to the caller.

  • When called party accepts the call, Call Agent sends 200 OK with yet
    another SDP (offer/answer?) points to the called party.

  • After that there is no voice path
    to be exact, voice path is still
    established to the media server providing tones but not to the called party
    as directed in the latter SDP.

Can you help me

Thx a lot

G

2

What was the sequence number on the original SDP and what is the new sequence number? Better: provide the actual SDP.

3

CSeq: 24564 INVITE for the two sdp.

4

The SDP version number (which must be greater in the second one).

5

here are the 2 SIP messages

Session Initiation Protocol (180)
    Status-Line: SIP/2.0 180 Ringing
        Status-Code: 180
        [Resent Packet: False]
        [Request Frame: 353]
        [Response Time (ms): 18]
    Message Header
        Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, REFER, SUBSCRIBE, OPTIONS, UPDATE
        Contact: sip:10.100.231.100
        User-Agent: OmniPCX Enterprise R12.1 m2.300.13.i
        Content-Type: application/sdp
        To: <sip:80800100@10.100.231.100>;tag=91d5ad07c9bb45beecd6688c9b1a2ded
        From: "INTERPHONE AJ_PR:2" <sip:80800402@10.100.231.100>;tag=5rGm287SGncxOk.YI3bvTyL1pcF9TVKl
        Call-ID: iuJ0UWtapSFQqIKqeMU5TCh28.A-zzME
        CSeq: 24564 INVITE
        Via: SIP/2.0/UDP 10.230.36.31:5060;received=10.230.36.31;rport=5060;branch=z9hG4bKPjo3Zar0q3rDjjWF9zDQKRcfrVU9chsAZ6
        Content-Length: 275
    Message Body
        Session Description Protocol
            Session Description Protocol Version (v): 0
            Owner/Creator, Session Id (o): OXE 1551715096 1551715096 IN IP4 10.100.231.100
            Session Name (s): abs
            Connection Information (c): IN IP4 10.230.36.2
            Time Description, active time (t): 0 0
            Media Description, name and address (m): audio 32556 RTP/AVP 8 101
            Media Attribute (a): sendrecv
            Media Attribute (a): rtpmap:8 PCMA/8000
            Media Attribute (a): rtpmap:101 telephone-event/8000
            Media Attribute (a): ptime:20
            Media Attribute (a): maxptime:30
            Media Description, name and address (m): video 0 RTP/AVP 99
            Media Attribute (a): rtpmap:99 H264/90000
            Media Attribute (a): sendrecv



Session Initiation Protocol (200)
    Status-Line: SIP/2.0 200 OK
        Status-Code: 200
        [Resent Packet: False]
        [Request Frame: 353]
        [Response Time (ms): 2876]
    Message Header
        Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, REFER, SUBSCRIBE, OPTIONS, UPDATE
        Contact: sip:10.100.231.100
        Supported: replaces,timer,path,100rel
        User-Agent: OmniPCX Enterprise R12.1 m2.300.13.i
        Session-Expires: 1800;refresher=uas
        P-Asserted-Identity: "xxxxxxxxx" <sip:80800100@10.100.231.100;user=phone>
        Content-Type: application/sdp
        To: <sip:80800100@10.100.231.100>;tag=91d5ad07c9bb45beecd6688c9b1a2ded
        From: "INTERPHONE AJ_PR:2" <sip:80800402@10.100.231.100>;tag=5rGm287SGncxOk.YI3bvTyL1pcF9TVKl
        Call-ID: iuJ0UWtapSFQqIKqeMU5TCh28.A-zzME
        CSeq: 24564 INVITE
        Via: SIP/2.0/UDP 10.230.36.31:5060;received=10.230.36.31;rport=5060;branch=z9hG4bKPjo3Zar0q3rDjjWF9zDQKRcfrVU9chsAZ6
        Content-Length: 277
    Message Body
        Session Description Protocol
            Session Description Protocol Version (v): 0
            Owner/Creator, Session Id (o): OXE 1551715096 1551715096 IN IP4 10.100.231.100
            Session Name (s): abs
            Connection Information (c): IN IP4 10.230.36.101
            Time Description, active time (t): 0 0
            Media Description, name and address (m): audio 32514 RTP/AVP 8 101
            Media Attribute (a): sendrecv
            Media Attribute (a): rtpmap:8 PCMA/8000
            Media Attribute (a): rtpmap:101 telephone-event/8000
            Media Attribute (a): ptime:20
            Media Attribute (a): maxptime:30
            Media Description, name and address (m): video 0 RTP/AVP 99
            Media Attribute (a): rtpmap:99 H264/90000
            Media Attribute (a): sendrecv
6

They are both 1551715096, so the second one is, correctly, being ignored as a duplicate.

chan_sip has a special option to correct for peers that are brain damaged in this way. I don’t know if there is something similar in PJSIP.

1 Like
7

Thx a lot David :slight_smile:
What is the option chan_sip?

8

The chan_sip option is ignoresdpversion