5 Jan
2017
5 Jan
'17
12:16 p.m.
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
<sip%3Ahuynhngocphap(a)happy.anttel-pro.ab-kz-02.antbuddy.com>>;tag=zJBNvD67y3E.1I5Y5ZrRI4JmP5JKeNWO#015#012To015#012To:
<sip:buiduchahai@happy.anttel-pro.ab-kz-02.antbuddy.com
<sip%3Abuiduchahai(a)happy.anttel-pro.ab-kz-02.antbuddy.com>>;tag=2SF4D790Zy6Kj#015#012Call-ID12Call-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/8000a=rtpmap:101 telephone-event/8000a=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:huynhngocphap@happy.anttel-pro.ab-kz-02.antbuddy.com
<sip%3Ahuynhngocphap(a)happy.anttel-pro.ab-kz-02.antbuddy.com>>;tag=zJBNvD67y3E.1I5Y5ZrRI4JmP5JKeNWO*
*To: <sip:buiduchahai@happy.anttel-pro.ab-kz-02.antbuddy.com
<sip%3Abuiduchahai(a)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