Hi all, You have any idea ? Thank you very much in advance!
Regards, Hai Bui On Fri, Jan 6, 2017 at 9:39 AM, Hai Bui Duc Ha <hai....@htklabs.com> 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: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. > <sip%3Ahuynhngocphap@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=z9hG4bKPjaXcUF2ZkxyQGYCb3a57cP > Q3rkcKMY.eS;alias > Max-Forwards: 50 > From: "Phap Huynh" <sip:huynhngocphap@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 > -- Hai Bui VoIP engineer, Cvoice team, HTK-HCM Office Mobile: +84-165-618-9876
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