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:buiducha...@happy.anttel-pro.ab-kz-02.antbuddy.com <sip%3abuiducha...@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:huynhngocp...@happy.anttel-pro.ab-kz-02.antbuddy.com <sip%3ahuynhngocp...@happy.anttel-pro.ab-kz-02.antbuddy.com>>;tag=zJBNvD67y3E.1I5Y5ZrRI4JmP5JKeNWO* * To: <sip:buiducha...@happy.anttel-pro.ab-kz-02.antbuddy.com <sip%3abuiducha...@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 <http://happy.anttel-pro.ab-kz-02.antbuddy.com>", nonce="452a7bce-d326-11e6-a605-e9dce514db6e", uri="sip:buiducha...@happy.anttel-pro.ab-kz-02.antbuddy.com <sip%3abuiducha...@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:buiducha...@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:huynhngocp...@happy.anttel-pro.ab-kz-02.antbuddy.com >;tag=zJBNvD67y3E.1I5Y5ZrRI4JmP5JKeNWO To: <sip:buiducha...@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:buiducha...@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 <mico...@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:huynhngocp...@happy.anttel-pro.ab-kz-02.antbuddy.com > <sip%3ahuynhngocp...@happy.anttel-pro.ab-kz-02.antbuddy.com>>;tag=zJBNvD67y3E.1I5Y5ZrRI4JmP5JKeNWO#015#012To: > <sip:buiducha...@happy.anttel-pro.ab-kz-02.antbuddy.com > <sip%3abuiducha...@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 > <http://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) > *ACK sip:buiduchahai@125.212.212.36:11000;transport=tcp SIP/2.0* > *Via: SIP/2.0/TCP > 10.0.2.15:57735;rport;branch=z9hG4bKPjlgc13AjrUrFJHq60vWhGqsUaGXi2F98Z;alias* > *Max-Forwards: 70* > *From: "Phap Huynh" > <sip:huynhngocp...@happy.anttel-pro.ab-kz-02.antbuddy.com > <sip%3ahuynhngocp...@happy.anttel-pro.ab-kz-02.antbuddy.com>>;tag=zJBNvD67y3E.1I5Y5ZrRI4JmP5JKeNWO* > *To: <sip:buiducha...@happy.anttel-pro.ab-kz-02.antbuddy.com > <sip%3abuiducha...@happy.anttel-pro.ab-kz-02.antbuddy.com>>;tag=2SF4D790Zy6Kj* > *Call-ID: ftMudIpIQeKWwP8kQDi2z1S0D1sV3KaB* > *CSeq: 29055 ACK* > *Route: > <sip:125.212.212.40;transport=tcp;lr;ftag=zJBNvD67y3E.1I5Y5ZrRI4JmP5JKeNWO>* > *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 > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing > listsr-us...@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > -- > Daniel-Constantin Mierlawww.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
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