Hi Daniel,
Thank you for reply.
On Tue, Jan 17, 2017 at 6:05 PM, Daniel-Constantin Mierla <miconda@gmail.com
wrote:
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 send you the pcap at enclosed file. You can see the packet *No.5 *, it missing SIP message body: * Media Attribute (a): rtpmap:8 PCMA/8000* * Media Attribute (a): rtpmap:101 telephone-event/8000* * Media Attribute (a): fmtp:101 0-16*
I expect that content length is mismatching or there is a '\0' inside the sdp.
Can you explain me more about this ?
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%3Abuiduchahai@happy.anttel-pro.ab-kz-02.antbuddy.com SIP/2.0*
- Message Header*
Via: SIP/2.0/TCP10.0.2.15:57735;rport;branch=z9hG4bKPjaXcUF2ZkxyQGYCb3a57cPQ3rkcKMY.eS;alias*
Max-Forwards: 70*From: "Phap Huynh"<sip:huynhngocphap@happy.anttel-pro.ab-kz-02.antbuddy.com sip%3Ahuynhngocphap@happy.anttel-pro.ab-kz-02.antbuddy.com>;tag=zJBNvD67y3E.1I5Y5ZrRI4JmP5JKeNWO*
To: <sip:buiduchahai@happy.anttel-pro.ab-kz-02.antbuddy.comsip%3Abuiduchahai@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:buiduchahai@happy.anttel-pro.ab-kz-02.antbuddy.com sip%3Abuiduchahai@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 IP410.0.2.15*
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 90 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=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: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 sip%3Ahuynhngocphap@happy.anttel-pro.ab-kz-02.antbuddy.com>;tag=zJBNvD67y3E.1I5Y5ZrRI4JmP5JKeNWO#015#012To: <sip:buiduchahai@happy.anttel-pro.ab-kz-02.antbuddy.com sip%3Abuiduchahai@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:huynhngocphap@happy.anttel-pro.ab-kz-02.antbuddy.com sip%3Ahuynhngocphap@happy.anttel-pro.ab-kz-02.antbuddy.com>;tag=zJBNvD67y3E.1I5Y5ZrRI4JmP5JKeNWO* *To: <sip:buiduchahai@happy.anttel-pro.ab-kz-02.antbuddy.com sip%3Abuiduchahai@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-users@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/cg i-bin/mailman/listinfo/sr-users
-- Hai Bui VoIP engineer, Cvoice team, HTK-HCM Office Mobile: +84-165-618-9876
-- Daniel-Constantin Mierlawww.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com
Regrads, Hai Bui