Hello,

apparently I missed the follow ups on this discussion, dragged in by other topics on mailing list.

Can you get the pcap with all the traffic taken on kamailio server for the call (from initial invite to the end of the call)? I expect that content length is mismatching or there is a '\0' inside the sdp.


On 06/01/2017 03:39, Hai Bui Duc Ha wrote:
Hello Daniel,

Thank for reply !

do you have the pcap for such message?

Here is the message, I capture via Wireshark on client:
Session Initiation Protocol (INVITE)
    Request-Line: INVITE sip:buiduchahai@happy.anttel-pro.ab-kz-02.antbuddy.com SIP/2.0
    Message Header
        Via: SIP/2.0/TCP 10.0.2.15:57735;rport;branch=z9hG4bKPjaXcUF2ZkxyQGYCb3a57cPQ3rkcKMY.eS;alias
        Max-Forwards: 70
        From: "Phap Huynh" <sip:huynhngocphap@happy.anttel-pro.ab-kz-02.antbuddy.com>;tag=zJBNvD67y3E.1I5Y5ZrRI4JmP5JKeNWO
        To: <sip:buiduchahai@happy.anttel-pro.ab-kz-02.antbuddy.com>
        Contact: <sip:huynhngocphap@49.156.54.54:50785;transport=TCP;ob>
        Call-ID: ftMudIpIQeKWwP8kQDi2z1S0D1sV3KaB
        CSeq: 29055 INVITE
        Route: <sip:125.212.212.40;transport=tcp;lr>
        Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
        Supported: replaces, 100rel, timer, norefersub
        Session-Expires: 1800
        Min-SE: 9
        User-Agent: CSipSimple_hlteatt-19/r55
         [truncated]Proxy-Authorization: Digest username="huynhngocphap", realm="happy.anttel-pro.ab-kz-02.antbuddy.com", nonce="452a7bce-d326-11e6-a605-e9dce514db6e", uri="sip:buiduchahai@happy.anttel-pro.ab-kz-02.antbuddy.com", response="71749de
        Content-Type: application/sdp
        Content-Length:   299
    Message Body
        Session Description Protocol
            Session Description Protocol Version (v): 0
            Owner/Creator, Session Id (o): - 3692596134 3692596134 IN IP4 10.0.2.15
            Session Name (s): pjmedia
            Connection Information (c): IN IP4 10.0.2.15
            Time Description, active time (t): 0 0
            Media Description, name and address (m): audio 4002 RTP/AVP 9 0 8 101
            Connection Information (c): IN IP4 10.0.2.15
            Media Attribute (a): rtcp:4003 IN IP4 10.0.2.15
            Media Attribute (a): sendrecv
            Media Attribute (a): rtpmap:9 G722/8000
            Media Attribute (a): rtpmap:0 PCMU/8000
            Media Attribute (a): rtpmap:8 PCMA/8000
            Media Attribute (a): rtpmap:101 telephone-event/8000
            Media Attribute (a): fmtp:101 0-16

And this is message I receive on Freeswitch:
 ------------------------------------------------------------------------
   INVITE sip:buiduchahai@happy.anttel-pro.ab-kz-02.antbuddy.com SIP/2.0
   Record-Route: <sip:125.212.212.40;transport=tcp;lr=on;ftag=zJBNvD67y3E.1I5Y5ZrRI4JmP5JKeNWO>
   Via: SIP/2.0/TCP 125.212.212.40:5060;branch=z9hG4bK0d89.3807a4cf41ce9b48a7d1a75826762d6e.0;i=533c
   Via: SIP/2.0/TCP 10.0.2.15:57735;received=49.156.54.54;rport=50785;branch=z9hG4bKPjaXcUF2ZkxyQGYCb3a57cPQ3rkcKMY.eS;alias
   Max-Forwards: 50
   From: "Phap Huynh" <sip:huynhngocphap@happy.anttel-pro.ab-kz-02.antbuddy.com>;tag=zJBNvD67y3E.1I5Y5ZrRI4JmP5JKeNWO
   To: <sip:buiduchahai@happy.anttel-pro.ab-kz-02.antbuddy.com>
   Contact: <sip:huynhngocphap@49.156.54.54:50785;transport=TCP;ob>
   Call-ID: ftMudIpIQeKWwP8kQDi2z1S0D1sV3KaB
   CSeq: 29055 INVITE
   Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
   Supported: replaces, 100rel, timer, norefersub
   Session-Expires: 1800
   Min-SE: 9
   User-Agent: CSipSimple_hlteatt-19/r55
   Proxy-Authorization: Digest username="huynhngocphap", realm="happy.anttel-pro.ab-kz-02.antbuddy.com", nonce="452a7bce-d326-11e6-a605-e9dce514db6e", uri="sip:buiduchahai@happy.anttel-pro.ab-kz-02.antbuddy.com", response="71749de35a220aef0b92c9ade03f90b7", algorithm=MD5, cnonce="px.OiQUg2zZmYh.0-MmLC0f.-ZPXYa1V", qop=auth, nc=00000001
   Content-Type: application/sdp
   Content-Length:   289
   X-AUTH-IP: 49.156.54.54
   X-AUTH-PORT: 50785

   v=0
   o=- 3692596134 3692596134 IN IP4 49.156.54.54
   s=pjmedia
   c=IN IP4 49.156.54.54
   t=0 0
   m=audio 4002 RTP/AVP 9 0 8 101
   c=IN IP4 49.156.54.54
   a=rtcp:4003
   a=sendrecv
   a=rtpmap:9 G722/8000
   a=rtpmap:0 PCMU/8000
   a=oldmediaip:10.0.2.15
   a=oldmediaip:10.0.2.15
   a=oldmediaip:10.0.2.15
   ------------------------------------------------------------------------

You can see, it missing:
            Media Attribute (a): rtpmap:8 PCMA/8000
            Media Attribute (a): rtpmap:101 telephone-event/8000
            Media Attribute (a): fmtp:101 0-16

 Is it happening for the ACK you pasted or some other message?
It only happen on ACK messages, when my client reply 200 OK /SDP to server to establish call.

For more detail: I use another soft phone on Android like Zoiper and test with the same scenario, it work ok. And when my client use 3G, it still work ok.

Regards,
Hai Bui



On Thu, Jan 5, 2017 at 10:25 PM, Daniel-Constantin Mierla <miconda@gmail.com> wrote:

Hello,

do you have the pcap for such message? Is it happening for the ACK you pasted or some other message?

Cheers,
Daniel


On 05/01/2017 12:16, Hai Bui Duc Ha wrote:
Hi all,

I have problem when make call with my Android mobile use PJSIP library.
Scenario:

my client  -> Kamailio -> Freeswitch (media server) -> another client (soft phone on Windows)

my client:
 + use Bluestack
 + Capture via Wireshark
 + use Wifi

Issue: The call will be drop after ~ 30 second.

I see the error on Kamailio:
Jan  5 16:08:59 ab-kz-02 kamailio[6343]: ERROR: <core> [parser/parse_fline.c:257]: parse_first_line(): parse_first_line: bad message (offset: 13)
Jan  5 16:08:59 ab-kz-02 kamailio[6343]: ERROR: <core> [parser/parse_fline.c:257]: parse_first_line(): parse_first_line: bad message (offset: 13)
Jan  5 16:08:59 ab-kz-02 kamailio[6343]: ERROR: <core> [parser/msg_parser.c:690]: parse_msg(): ERROR: parse_msg: message=<p:8 PCMA/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-16#015#012ACK sip:buiduchahai@125.212.212.36:11000;transport=tcp SIP/2.0#015#012Via: SIP/2.0/TCP 10.0.2.15:57735;rport;branch=z9hG4bKPjlgc13AjrUrFJHq60vWhGqsUaGXi2F98Z;alias#015#012Max-Forwards: 70#015#012From: "Phap Huynh" <sip:huynhngocphap@happy.anttel-pro.ab-kz-02.antbuddy.com>;tag=zJBNvD67y3E.1I5Y5ZrRI4JmP5JKeNWO#015#012To: <sip:buiduchahai@happy.anttel-pro.ab-kz-02.antbuddy.com>;tag=2SF4D790Zy6Kj#015#012Call-ID: ftMudIpIQeKWwP8kQDi2z1S0D1sV3KaB#015#012CSeq: 29055 ACK#015#012Route: <sip:125.212.212.40;transport=tcp;lr;ftag=zJBNvD67y3E.1I5Y5ZrRI4JmP5JKeNWO>#015#012Content-Length:  0#015#012#015#012>
Jan  5 16:08:59 ab-kz-02 kamailio[6343]: ERROR: <core> [receive.c:129]: receive_msg(): core parsing of SIP message failed (49.156.54.54:50785/2)

Seem to the server error when parse
(on INVITE SDP)
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

(on new ACK message)
Via: SIP/2.0/TCP 10.0.2.15:57735;rport;branch=z9hG4bKPjlgc13AjrUrFJHq60vWhGqsUaGXi2F98Z;alias
Max-Forwards: 70
From: "Phap Huynh" <sip:huynhngocphap@happy.anttel-pro.ab-kz-02.antbuddy.com>;tag=zJBNvD67y3E.1I5Y5ZrRI4JmP5JKeNWO
Call-ID: ftMudIpIQeKWwP8kQDi2z1S0D1sV3KaB
CSeq: 29055 ACK
Content-Length:  0

I think the SIP message is fragmented but when resume package is not correct.
Do you have any advice ? Thank you for watching !

Regards,
Hai Bui


--
Hai Bui
VoIP engineer, Cvoice team, HTK-HCM Office
Mobile: +84-165-618-9876


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-- 
Daniel-Constantin Mierla
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Hai Bui
VoIP engineer, Cvoice team, HTK-HCM Office
Mobile: +84-165-618-9876 
-- 
Daniel-Constantin Mierla
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com