[asterisk-bugs] [JIRA] (ASTERISK-24889) Possible bug: PJSIP: stateless behavior when transport=tcp

Sat Mar 21 18:17:34 CDT 2015

    [ https://issues.asterisk.org/jira/browse/ASTERISK-24889?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=225541#comment-225541 ] 

Anatoli commented on ASTERISK-24889:
------------------------------------

Olle,

It looks like there are 2 problems involved in this issue. First one is what Joshua described, that res_pjsip won't reuse TCP connections if for the endpoint in question there is a {{transport=<transport_name>}} option specified in pjsip.conf.

But there is also another problem (as you suspected): the Contact header filed that the softphones send is (in my opinion) incorrectly formatted.

>From SIP RFP: _"While the Via header field tells other elements where to send the response, the Contact header field tells other elements where to send future requests."_

And this is what Jitsi on Windows 7 sends:

{code}REGISTER sip:server_ip;transport=tcp SIP/2.0
Call-ID: dd753e83b597b5ca2a2e4f162775375e at 0:0:0:0:0:0:0:0
CSeq: 1 REGISTER
From: "Name1" <sip:555 at server_ip>;tag=1dde9f02
To: "Name1" <sip:555 at server_ip>
Via: SIP/2.0/TCP host_ip:3409;branch=z9hG4bK-313537-071be23f01c7c2ee583675273d44e27d
Max-Forwards: 70
User-Agent: Jitsi2.6.5390Windows 7
Expires: 600
Contact: "Name1" <sip:555 at host_ip:3409;transport=tcp;registering_acc=10_101_10_1>;expires=600
Content-Length: 0{code}

Here Via says _"send the responses to host_ip:3409"_ and Contact says _"send new requests (INVITE, BYE), establishing a new TCP session, to.. the same host_ip:3409"_. If I understand it correctly, {{Contact}} should say {{host_ip:5060}} or omit the port entirely for the default value to be applied. Nevertheless Jitsi listens on all 3 standard SIP ports: tcp/5060, tcp/5061 & udp/5060 on 0.0.0.0 (they are specified in the global SIP configuration section) BUT the 3409 (random) port appears in netstat as ESTABLISHED, _not_ LISTENING and it doesn't accept new incoming connections.

So, Jitsi expects incoming connections on 5060, but specifies the incoming port as 3409 in {{Contact}} field.

Same behavior is observed in Linphone and SessionChat/SessionTalk both on iPhone:

{code}REGISTER sip:server_ip SIP/2.0
Via: SIP/2.0/TCP host2_ip:59911;alias;branch=z9hG4bK.3kRQgBAbb;rport
From: <sip:777 at server_ip>;tag=4Yzj~FqLz
To: sip:777 at server_ip
CSeq: 20 REGISTER
Call-ID: 47V~RpZbNs
Max-Forwards: 70
Supported: outbound
Contact: <sip:777 at host2_ip:59911;transport=tcp>;+sip.instance="<urn:uuid:1e603d84-0b11-42a7-83cb-7c34f71e0089>"
Expires: 600
User-Agent: LinphoneIphone/2.2.4.1-27-ge8ab36a (belle-sip/1.3.3)
Content-Length: 0{code}

{code}REGISTER sip:server_ip SIP/2.0
Via: SIP/2.0/TCP host2_ip:35210;branch=z9hG4bK-524287-1---0a74526f8a4d825c;rport;alias
Max-Forwards: 70
Contact: <sip:777 at host2_ip:35210;rinstance=2a2202a7c0b108dd;transport=tcp>
To: "Name1"<sip:777 at server_ip>
From: "Name1"<sip:777 at server_ip>;tag=02dd3253
Call-ID: QFlg48EH71b0n3hklDy1IQ..
CSeq: 1 REGISTER
Expires: 600
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, SUBSCRIBE
Supported: replaces, norefersub
User-Agent: SessionTalk Version 5.11
Content-Length: 0{code}

A similar discussion is held in the pjsip dev mailing list: http://comments.gmane.org/gmane.comp.voip.pjsip/19202.

> Possible bug: PJSIP: stateless behavior when transport=tcp
> ----------------------------------------------------------
>
>                 Key: ASTERISK-24889
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24889
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: pjproject/pjsip
>    Affects Versions: 13.2.0
>         Environment: OS: Ubuntu 14.04, with all updates
> Kernel: 3.13.0-44-generic
> pjsip: 2.3 from http://www.pjsip.org/release/2.3/pjproject-2.3.tar.bz2 (./configure --prefix=/usr --enable-shared --enable-epoll; make; sudo make install)
> Asterisk: 13.2.0 from http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-13.2.0.tar.gz (./configure --prefix=/usr --sysconfdir=/etc --localstatedir=/var; make; sudo make install) 
>            Reporter: Anatoli
>
> When an endpoint registers via TCP, PJSIP processes everything as corresponds: the communication is via TCP, REGISTER/ACK pass OK, the endpoint can even establish calls. But, when another endpoint calls to the first one, PJSIP sends INVITE message in a new TCP session to the already opened port.
> So if the registration was performed from TCP 55880 to 5060 and everything except INVITE flows in the same TCP session, INVITE is sent via a new TCP session from a random port on asterisk server (e.g. 32926) to the already opened 55880 port. With tcpdump I see a new SYN packet (and the already established TCP session was not closed/reset yet):
> {noformat}
> From: 10.101.10.1 (asterisk IP)
> To: 10.101.100.100 (endpoint IP)
> Source port: 32926
> Destination port: 55880
> Protocol: TCP
> TCP Data: [SYN] Seq=0 Win=29200 Len=0 MSS=1460 SACK_PERM=1 TSval=9227246 TSecr=0 WS=128
> {noformat}
> Naturally, the endpoint responds with a RST,ACK.
> The CLI shows the following:
> {noformat}
> ERROR[19352]: pjsip:0 <?>:    tcpc0x7fcdb404 TCP connect() error: Connection refused [code=120111]
> WARNING[19352]: pjsip:0 <?>:  tsx0x7fcdb402e Failed to send Request msg INVITE/cseq=31054 (tdta0x7fcdb4034260)! err=120111 (Connection refused)
> {noformat}
> Investigating the code, I've found that to send the INVITE message PJSIP calls pjsip_endpt_send_request_stateless, which, as the name indicates, behaves statelessly, though using TCP.
> Call trace:
> {noformat}
> sip_dialog.c at 1164 pjsip_dlg_send_request(INVITE) => @1214 call to pjsip_tsx_create_uac
> sip_transaction.c at 1266 pjsip_tsx_create_uac  => call to pjsip_tsx_create_uac2
> sip_transaction.c at 1273 pjsip_tsx_create_uac2  => @1367 tsx_on_state_null  callback assignment
> sip_transaction.c at 2311 tsx_on_state_null => @2351 call to tsx_send_msg
> sip_transaction.c at 2006 tsx_send_msg => @2120 call to pjsip_endpt_send_request_stateless
> {noformat}
> Asterisk 13.2.0, PJSIP 2.3, everything compiled from sources.
> Config: same as in the examples except for one endpoint the transport=<tcp_transport> option is specified.
> Client: a number of softphones, tested with latest Linphone on iPhone, SessionChat (freeware from AppStore), Jitsi on Windows.
> Please let me know if additional information is required.
> Regards,
> Anatoli

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