[asterisk-bugs] [JIRA] (ASTERISK-24807) Missing mandatory field Max-Forwards

[Anatoli (JIRA)]

Anatoli created ASTERISK-24807:
----------------------------------

             Summary: Missing mandatory field Max-Forwards
                 Key: ASTERISK-24807
                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24807
             Project: Asterisk
          Issue Type: Bug
      Security Level: None
          Components: pjproject/pjsip
    Affects Versions: 13.2.0
         Environment: OS: Ubuntu 12.04 inside VirtualBox, with all updates

Kernel: 3.8.0-29-generic

pjsip: 2.3 from http://www.pjsip.org/release/2.3/pjproject-2.3.tar.bz2 (./configure --prefix=/usr --enable-shared --enable-epoll --with-external-srtp; 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
            Severity: Critical


While evaluating the latest Asterisk and PJSIP (both compiled from sources), when calling to/from softphones (in particular the latest Linphone v.3.7.0) the INVITE messages from Asterisk are rejected due to the specification violation.

Here are more people having the same problem: http://comments.gmane.org/gmane.comp.telephony.pbx.asterisk.user/281228 and http://lists.digium.com/pipermail/asterisk-dev/2003-July/001114.html.

Linphone has a debug option: {{linphone.exe --logfile "c:\Temp\logs.txt"}} and in the generated logfile one can see something like:

{code}[error] Missing mandatory header [Max-Forwards] for message [INVITE]{code}

\\
And this is what Asterisk sends to it:

{code}
INVITE sip:6002 at 192.168.56.102 SIP/2.0
Via: SIP/2.0/UDP 192.168.56.3:5060;rport;branch=z9hG4bKPjbf53c93c-a424-4b9e-9e6a-b50d0e606965
From: <sip:6001 at 192.168.56.3>;tag=deeff9b9-50ab-410e-b683-94fdeb9f87b3
To: <sip:6002 at 192.168.56.102>
Contact: <sip:9426fd27-7ccd-4e1e-ab7a-e6c27fc59a5c at 192.168.56.3:5060>
Call-ID: 5b517cda-634b-4036-bb81-95a43379c09d
CSeq: 6848 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Content-Type: application/sdp
Content-Length:   254
{code}

So indeed, there is no "Max-Forwards" header sent with the INVITE requests, but, according to the SIP RFC 3261: "A valid SIP request formulated by a UAC MUST, at a minimum, contain the following header fields: To, From, CSeq, Call-ID, Max-Forwards, and Via; all of these header fields are mandatory in all SIP requests."
\\
After researching a bit the source code of pjsip, I found that actually almost all messages originating from it don't have Max-Forwards fields.

I've tried to add the header manually and it solves the problem (this is not a definitive solution; it has some minor issues):

{code}
--- pjproject-2.3/pjsip/src/pjsip/sip_util.c.orig  2014-08-18 05:54:43.000000000 -0300
+++ pjproject-2.3/pjsip/src/pjsip/sip_util.c        2015-02-17 10:52:23.839673699 -0300
@@ -465,6 +465,8 @@ PJ_DEF(pj_status_t) pjsip_endpt_create_r
     }
     PJ_END;

+             pjsip_msg_add_hdr(tdata->msg, pjsip_max_fwd_hdr_create(tdata->pool, 70)); 
+
     *p_tdata = tdata;
     return PJ_SUCCESS;
{code}

But then I've discussed it with people at pjsip at lists.pjsip.org (here is the thread: http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/2015-February/018120.html) and they say that the issue is actually within the code that integrates pjsip with asterisk (e.g. {{res/res_pjsip/config_global.c}}) and suggested filling an issue here.

\\
This problem is a show-stopper when there are softphones among the endpoints.

----

STEPS TO REPRODUCE

*extensions.conf*

{code}
[general]
language=en
static=yes
writeprotect=no
autofallthrough=yes

[internal]
exten => 6001,1,Answer()
same => 2,Dial(PJSIP/6001,20)
same => 3,Hangup()

exten => 6002,1,Answer()
same => 2,Dial(PJSIP/6002,20)
same => 3,Hangup()
{code}



*pjsip.conf*

{code}
[transport-udp]
type=transport
protocol=udp
bind=192.168.56.3


[6001]
type=endpoint
transport=transport-udp
context=internal
allow=ulaw
allow=gsm
auth=6001
aors=6001

device_state_busy_at=1
allow_subscribe=yes
sub_min_expiry=30

[6001]
type=auth
auth_type=userpass
password=6001
username=6001

[6001]
type=aor
max_contacts=1
contact=sip:6001 at 192.168.56.101


[6002]
type=endpoint
transport=transport-udp
context=internal
allow=ulaw
allow=gsm
auth=6002
aors=6002

device_state_busy_at=1
allow_subscribe=yes
sub_min_expiry=30

[6002]
type=auth
auth_type=userpass
password=6002
username=6002


[6002]
type=aor
max_contacts=1
contact=sip:6002 at 192.168.56.102
{code}


Start asterisk:
{{# /usr/sbin/asterisk}}


Then configure the extensions in 2 Linphones (in wizard: user/pass =
600x/600x and domain: 192.168.56.3) and try to make a call from one Linphone to another.



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