Hi Harold,
Are you sure your UA is not adding this header somehow outside the pjsip code? If I understand it correctly (and I may perfectly not as I just started analyzing SIP internals and pjsip code when I found this problem 2 days ago) pjsip handles almost everything SIP-related in asterisk by itself without much intervention from asterisk.
I've also continued pjsip code research a bit more and (I guess) I can confirm that the function where all outgoing requests from pjsip/asterisk originate is pjsip_endpt_create_request_from_hdr. I created a better patch (than the one I've sent with the first mail) that could be an acceptable workaround for everyone who encounters the same problem until a definitive fix is implemented. Here it is:
--- 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;
To apply it (for those who don't know how), cd to pjproject-2.3 source files folder, and there:
patch -p1 < <path_to_the_patch>/max_fwd.patch
This patch doesn't implement a proper decrement of max_fwd field as, according to SIP RFC, when relaying a request, max_fwd should be decremented by 1 and if it reaches 0, the request should be dropped to avoid infinite relaying loops (the same way TTL works in IP packets). Also it's initialized with a hardcoded (70) value but should be taking it from [general] max_forwards option.
But, nevertheless, with this patch I haven't found any problems using 2 (Linphone) softphones, so my guess is that this functionality is part of pjsip rather than asterisk, but again I may be completely wrong. I guess we could ask for a comment about this on asterisk-dev mailing list.
Regards,
Anatoli
From: pjsip [mailto:pjsip-bounces@xxxxxxxxxxxxxxx] On Behalf Of Harald Radke
Sent: Wednesday, February 18, 2015 06:05
To: pjsip at lists.pjsip.org
Subject: Re: Possible bug - missing mandatory field Max-Forwards
Hi Anatoli,
Hm, right now this looks to me more like an asterisk than an pjsip problem.
Over here, my user agent creates headers with the Max-Forwards field, so I take it the bug
is more in the asterisk module that is using PJSIP than the lib itself.
Regards,
Harry
Gesendet: Mittwoch, 18. Februar 2015 um 02:35 Uhr
Von: Anatoli <me at anatoli.ws>
An: pjsip at lists.pjsip.org
Betreff: [pjsip] Possible bug - missing mandatory field Max-Forwards
Hi All,
I was trying to make work the latest asterisk (13.2.0) with the latest
pjproject (2.3) both compiled from sources. They work pretty well, but 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:
[error] Missing mandatory header [Max-Forwards] for message [INVITE]
And this is what pjsip at asterisk sends to it:
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@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
So indeed, there is no "Max-Forwards" header send 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.
As a proof-of-concept, here is a patch that adds this header to the INVITE
and ACK packets and with it the softphones work almost well. Almost, because
there are more types of packets involved in a communication (like hanging up
that doesn't work) and I've just patched 2 of them. Also, this patch doesn't
decrease the value of the header as it should to when the message is relayed
(Max-Forwards in SIP is like TTL field in IP), it's just a proof-of-concept.
IMO, the better way would be to modify accordingly the functions like
pjsip_endpt_create_request_from_hdr where the messages are created initially
and where other mandatory fields are set, but it requires a better
understanding of the project's code and SIP protocol in general than mine is
(it's almost 0). As a side note, it would be great to have the initial
Max-Forwards value configurable with the (already existing, but ignored)
max_forwards option in [general] section.
This problem is actually a show-stopper when there are softphones among the
endpoints; so as for me for example, I have to stay with the old asterisk
sip driver until this issue is solved.
Regards,
Anatoli
A *demo* patch:
--- pjproject-2.3/pjsip/src/pjsip-ua/sip_inv.c.orig 2014-08-21
04:20:34.000000000 -0300
+++ pjproject-2.3/pjsip/src/pjsip-ua/sip_inv.c 2015-02-16
14:55:29.683566529 -0300
@@ -1679,6 +1679,15 @@ PJ_DEF(pj_status_t) pjsip_inv_invite( pj
}
}
+
+
+ // PATCH
+ if (!pjsip_msg_find_hdr(tdata->msg, PJSIP_H_MAX_FORWARDS, NULL))
+ pjsip_msg_add_hdr(tdata->msg,
pjsip_max_fwd_hdr_create(tdata->pool, 70));
+ // PATCH
+
+
+
/* See if we have SDP to send. */
if (inv->neg) {
pjmedia_sdp_neg_state neg_state;
@@ -2986,6 +2995,12 @@ PJ_DEF(pj_status_t) pjsip_inv_create_ack
return status;
}
+
+ // PATCH
+ pjsip_msg_add_hdr(inv->last_ack->msg,
pjsip_max_fwd_hdr_create(inv->last_ack->pool, 70));
+ // PATCH
+
+
/* See if we have pending SDP answer to send */
sdp = inv_has_pending_answer(inv, inv->invite_tsx);
if (sdp) {
STEPS TO REPRODUCE:
extensions.conf:
[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()
pjsip.conf:
[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
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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