[OpenSIPS-Devel] [OpenSIPS/opensips] a45d4d: nathelper: manual backport for 3c39167e333a8801772...

Maxim Sobolev sobomax at sippysoft.com
Mon Feb 5 15:24:37 EST 2018 

P.P.S. Our test case is quite simple you can see it here:

https://github.com/sippy/voiptests/blob/master/opensips.cfg.in

The failure observed here is that the IP and port in the SDP are not
updated despite session has been established in the rtpproxy just fine.

245 00:00:02.433/DBUG:GLOBAL:get_command: received command "20954_11
Uc18,0,2,4,8,96,97,98,101 S<R)0Z2_i5*RbILdz4toBU(nND~AS"]n@?{}06%]3B>p7IFo[
153.135.123.216 10604 gFcf*kxs6wTleVZC*'%7ipApCKX%i.4u;1"
246 00:00:02.433/INFO:GLOBAL:rtpp_command_ul_handle: new IPv4/IPv4 session
S<R)0Z2_i5*RbILdz4toBU(nND~AS"]n@?{}06%]3B>p7IFo[, tag
gFcf*kxs6wTleVZC*'%7ipApCKX%i.4u;1 requested, type strong
247 00:00:02.433/INFO:S<R)0Z2_i5*RbILdz4toBU(nND~AS"]n@?{}06%]3B>p7IFo[:rtpp_command_ul_handle:
new session on IPv4 port 14696 created, tag
gFcf*kxs6wTleVZC*'%7ipApCKX%i.4u;1
248 00:00:02.433/INFO:S<R)0Z2_i5*RbILdz4toBU(nND~AS"]n@?{}06%]3B>p7IFo[:rtpp_stream_prefill_addr:
pre-filling caller's RTP address with 153.135.123.216:10604
249 00:00:02.433/INFO:S<R)0Z2_i5*RbILdz4toBU(nND~AS"]n@?{}06%]3B>p7IFo[:rtpp_stream_prefill_addr:
pre-filling caller's RTCP address with 153.135.123.216:10605
250 00:00:02.433/DBUG:GLOBAL:rtpc_doreply: sending reply "14696\n"

However message received by Bob:

      1 00:00:02.416/GLOBAL/bob_ua: RECEIVED message from 127.0.0.1:5060:
      2 INVITE sip:bob_reinv_fail at 127.0.0.1:5062 SIP/2.0
      3 Record-Route:
<sip:127.0.0.1;lr;ftag=gFcf*kxs6wTleVZC*'%7ipApCKX%i.4u>
      4 Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bKfb6d.53bce7a4.0
      5 Via: SIP/2.0/UDP 127.0.0.1:5061
;received=127.0.0.1;branch=z9hG4bKf93b3d20f9059b593d6f640656ba5590;rport=5061
      6 Max-Forwards: 69
      7 From: "Alice Smith" <sip:alice_reinv_fail_ipv4 at 127.0.0.1
>;tag=gFcf*kxs6wTleVZC*'%7ipApCKX%i.4u
      8 To: <sip:bob_reinv_fail at 127.0.0.1>
      9 Call-ID: S<R)0Z2_i5*RbILdz4toBU(nND~AS"]n@?{}06%]3B>p7IFo[
    10 CSeq: 200 INVITE
    11 Contact: Anonymous <sip:alice_reinv_fail_ipv4 at 127.0.0.1:5061>
    12 Expires: 300
    13 User-Agent: Sippy
    14 cisco-GUID: 1970357383-1076098593-3422611852-2102670268
    15 h323-conf-id: 1970357383-1076098593-3422611852-2102670268
    16 Content-Type: application/sdp
    17 Content-Length: 407
    18
    19 v=0
    20 o=- 423683522800 423683522800 IN IP4 127.0.0.1
    21 s=-
    22 c=IN IP4 153.135.123.216
    23 t=0 0
    24 m=audio 14696 RTP/AVP 18 0 2 4 8 96 97 98 101
    25 a=rtpmap:18 G729a/8000
    26 a=rtpmap:0 PCMU/8000
    27 a=rtpmap:2 G726-32/8000
    28 a=rtpmap:4 G723/8000
    29 a=rtpmap:8 PCMA/8000
    30 a=rtpmap:96 G726-40/8000
    31 a=rtpmap:97 G726-24/8000
    32 a=rtpmap:98 G726-16/8000
    33 a=rtpmap:101 telephone-event/8000
    34 a=fmtp:101 0-15
    35 a=ptime:30
    36 a=sendrecv

On Mon, Feb 5, 2018 at 12:07 PM, Maxim Sobolev <sobomax at sippysoft.com>
wrote:

> P.S. Build logs are here:
>
> https://travis-ci.org/sippy/voiptests/builds/337694673
>
> On Mon, Feb 5, 2018 at 12:06 PM, Maxim Sobolev <sobomax at sippysoft.com>
> wrote:
>
>> That merge broke all of our OpenSIPS test scenarous on voiptests. We are
>> investigating about the possible cause but I think this change may need to
>> be reverted from the stable branches until it's clear what's going on.
>>
>> -Maxim
>>
>> On Mon, Feb 5, 2018 at 10:04 AM, Ovidiu Sas <osas at voipembedded.com>
>> wrote:
>>
>>>   Branch: refs/heads/2.3
>>>   Home:   https://github.com/OpenSIPS/opensips
>>>   Commit: a45d4dcaf046bb273cfe5905ac035845a6867945
>>>       https://github.com/OpenSIPS/opensips/commit/a45d4dcaf046bb27
>>> 3cfe5905ac035845a6867945
>>>   Author: Ovidiu Sas <osas at voipembedded.com>
>>>   Date:   2018-02-05 (Mon, 05 Feb 2018)
>>>
>>>   Changed paths:
>>>     M modules/nathelper/nathelper.c
>>>
>>>   Log Message:
>>>   -----------
>>>   nathelper: manual backport for 3c39167e333a8801772f49f8561a01
>>> 5bfa1836f1
>>>  - fix test condition for updateing IP for fix_nated_sdp()
>>>
>>>
>>>
>>> _______________________________________________
>>> Devel mailing list
>>> Devel at lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
>>>
>>>
>>
>>
>> --
>> Maksym Sobolyev
>> Sippy Software, Inc.
>> Internet Telephony (VoIP) Experts
>> Tel (Canada): +1-778-783-0474 <(778)%20783-0474>
>> Tel (Toll-Free): +1-855-747-7779 <(855)%20747-7779>
>> Fax: +1-866-857-6942 <(866)%20857-6942>
>> 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 <(778)%20783-0474>
> Tel (Toll-Free): +1-855-747-7779 <(855)%20747-7779>
> Fax: +1-866-857-6942 <(866)%20857-6942>
> 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
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