[OpenSIPS-Devel] "Early cancel" issue in the tm module

[Maxim Sobolev]

OK, further investigation revealed that the lack of the provisional
reply is caused by the usage of the t_newtran() in our routing script,
which then causes different code path in the t_relay(), i.e.
t_forward_nonack() function is used instead of t_relay_to(). Which
appears to be triggering a separate bug in the OpenSIPS that affects
all releases out there. We use t_newtran() to  make sure each request
is processed only once, otherwise each re transmit would cause
separate request to the rtpproxy to be generated. I think the fix
would be to extend t_forward_nonack() to properly generate 100 Trying.

Just in case, our routing script looks like the following:

==========================
    ## shield us against retransmits
    if (!t_newtran()) {
        sl_reply_error();
        exit;
    };

    if (method == "INVITE") {
        if (rtpproxy_offer("r")) {
            t_on_reply("1");
            t_on_failure("1");
        };
    };

    if (method == "BYE") {
        unforce_rtp_proxy();
    };

    record_route();

    if (loose_route()) {
        t_relay();
        exit;
    };

    if (src_ip == "127.0.0.1" && src_port == 5061) {
        t_relay("udp:127.0.0.1:5062");
    } else {
        t_relay("udp:127.0.0.1:5061");
    };
==========================

Please note that Kamailio does not have neither issue with early
CANCEL + lost 100, nor this issue. For now I've resolved it by adding
explicit t_reply("100", "Trying").

On the positive note, you patch seems to be working, thanks. Let me
know if you want me to open issue wrt that t_newtran's 100 Trying
problem.

https://github.com/sippy/voiptests/commit/11a6bc6bee1ddfc83eb4acf784ce7aebf8e6815b

On Wed, Dec 23, 2015 at 5:05 AM, Maxim Sobolev <sobomax at sippysoft.com> wrote:
> Sorry, Bogdan, please disregard my previous message. After closer look
> at the log it seems that my initial diagnosis was not quite correct.
> The problem appears that provisional response generation is broken
> somehow, so that the originating UA keeps sending INVITEs over and
> over again and could not initiate outbound CANCEL at all until 180
> comes in.
>
> 00:00:07.297/GLOBAL/alice_ua: RECEIVED message from 127.0.0.1:5060:
> SIP/2.0 180 Ringing
>
> I am investigating as to what could be a problem, looks like this
> condition might have been hidden and just exposed now. It seems that
> it has nothing to do with the change in question.
>
> On Wed, Dec 23, 2015 at 4:20 AM, Maxim Sobolev <sobomax at sippysoft.com> wrote:
>> Hi Bogdan, that patch is not fully functional. Yes, it did restore
>> INVITE re-transmits, so that the second 100 Trying is stimulated
>> properly, but the CANCEL to the egress call leg is not generated until
>> 180 Ringing response comes in. You can see full log here:
>>
>> https://s3.amazonaws.com/archive.travis-ci.org/jobs/98496601/log.txt
>>
>> See Call-ID: a0f27d0745c1a6b52348900c67362be8 at 172.17.0.79 for an example:
>>
>>
>> On Mon, Dec 14, 2015 at 5:31 PM, Bogdan-Andrei Iancu
>> <bogdan at opensips.org> wrote:
>>> Maxim,
>>>
>>> Please check the attached patch for 1.11
>>>
>>> Regards,
>>>
>>> Bogdan-Andrei Iancu
>>> OpenSIPS Founder and Developer
>>> http://www.opensips-solutions.com
>>>
>>> On 14.12.2015 22:46, Maxim Sobolev wrote:
>>>>
>>>> Thanks Bogdan, I will be working on test case this week. I'll drop you
>>>> a note when it's ready.
>>>>
>>>> On Fri, Dec 11, 2015 at 8:42 AM, Bogdan-Andrei Iancu
>>>> <bogdan at opensips.org> wrote:
>>>>>
>>>>> Roger that, thanks for the detailed explanations - let's be consistent in
>>>>> handling the INVITE cancelling, in terms of letting all the time the
>>>>> final
>>>>> UAS to generate the final reply (if still alive).
>>>>>
>>>>> Best regards,
>>>>>
>>>>> Bogdan-Andrei Iancu
>>>>> OpenSIPS Founder and Developer
>>>>> http://www.opensips-solutions.com
>>>>>
>>>>>
>>>
>>
>>
>>
>> --
>> Maksym Sobolyev
>> Sippy Software, Inc.
>> Internet Telephony (VoIP) Experts
>> Tel (Canada): +1-778-783-0474
>> Tel (Toll-Free): +1-855-747-7779
>> Fax: +1-866-857-6942
>> Web: http://www.sippysoft.com
>> MSN: sales at sippysoft.com
>> Skype: SippySoft
>
>
>
> --
> Maksym Sobolyev
> Sippy Software, Inc.
> Internet Telephony (VoIP) Experts
> Tel (Canada): +1-778-783-0474
> Tel (Toll-Free): +1-855-747-7779
> Fax: +1-866-857-6942
> Web: http://www.sippysoft.com
> MSN: sales at sippysoft.com
> Skype: SippySoft



-- 
Maksym Sobolyev
Sippy Software, Inc.
Internet Telephony (VoIP) Experts
Tel (Canada): +1-778-783-0474
Tel (Toll-Free): +1-855-747-7779
Fax: +1-866-857-6942
Web: http://www.sippysoft.com
MSN: sales at sippysoft.com
Skype: SippySoft