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/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)

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>;tag=zJBNvD67y3E.1I5Y5ZrRI4JmP5JKeNWO

To: <sip:buiduchahai@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

VoIP engineer, Cvoice team, HTK-HCM Office

Mobile: +84-165-618-9876