[OpenSIPS-Users] SDP missing in ACK indialog reinvite only in 1.11.5
Louis Rochon
lrochon at solacom.com
Thu Mar 24 20:16:25 CET 2016
I recompiled with the 1.11.6, and did not resolve the issue. Traces show behaviour identical between 1.11.5 and 1.11.6.
Looking at the release notes for 1.11.6, there are some fixes that are closely related, but don't fix my specific issue. i.e. 95f5f79, 26a0a62.
Any suggestions? Is there some yet-to-be documented B2BLogic module parameter that needs to be set?
Louis
-----Original Message-----
From: users-bounces at lists.opensips.org [mailto:users-bounces at lists.opensips.org] On Behalf Of Louis Rochon
Sent: Monday, March 21, 2016 2:22 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] SDP missing in ACK indialog reinvite only in 1.11.5
Thanks Liviu,
I will recompile and give it a try.
Louis
-----Original Message-----
From: users-bounces at lists.opensips.org [mailto:users-bounces at lists.opensips.org] On Behalf Of Liviu Chircu
Sent: Monday, March 21, 2016 9:31 AM
To: users at lists.opensips.org
Subject: Re: [OpenSIPS-Users] SDP missing in ACK indialog reinvite only in 1.11.5
Hi Louis,
There was a similar discussion back in August [1], concluding with a fix [2], made between the 1.11.5 and 1.11.6 releases. Updating to the latest OpenSIPS 1.11 will most likely solve this problem.
[1]: http://lists.opensips.org/pipermail/users/2015-August/032239.html
[2]: https://github.com/OpenSIPS/opensips/commit/95f5f79b9250
Liviu Chircu
OpenSIPS Developer
http://www.opensips-solutions.com
On 21.03.2016 14:55, Louis Rochon wrote:
> Anybody?
>
> Louis
>
>
> -----Original Message-----
> From: users-bounces at lists.opensips.org
> [mailto:users-bounces at lists.opensips.org] On Behalf Of Louis Rochon
> Sent: Wednesday, March 16, 2016 10:02 AM
> To: users at lists.opensips.org
> Subject: [OpenSIPS-Users] SDP missing in ACK indialog reinvite only in
> 1.11.5
>
> SDP in ACK lost in OpenSIPS 1.11.5
>
> This is something that used to work in OpenSIPS 1.8.1, but seems to have been broken 1.11.5.
>
> Using the B2BUA facilities, I make the second leg of the call using b2b_init_request. Then, if required, I move the call to another user agent via a bash shell script, which issues a opensipsctl fifo b2b_bridge.
>
> For this in-dialog reinvite to work, there has to be an SDP in the ACK, which is missing the trace produced using OpenSIPS 1.11.5. It's there in 1.8.1.
>
> Anyway of reintroducing the SDP in the ACK in 1.11.5?
>
> Trace Explanation
> -----------------
> Objects: 10.10.8.103: source of initial call. Recipipient of reinvite.
> 10.10.10.205: CentOS running OpenSIPS with B2BUA. In one case, it's CentOS 5.11 running OpenSIPs 1.8.1, the other (broken) is CentOS 6.7 running OpenSIPs 1.11.5.
>
> Call Flow in Trace:
> 10.10.10.205 10.10.8.103
> --Invite--> Indialog reinvite, so SDP
> <--Trying--
> <--200 OK-- With SDP
> --ACK --> With SDP in 1.8.1, no SDP in 1.11.5
>
> OpenSIPS 1.8.1:
> ---------------
> INVITE sip:4507710011 at 10.10.8.103;user=phone SIP/2.0
> Via: SIP/2.0/UDP 10.10.10.205;branch=z9hG4bK109d.ce759b97.0
> To: "4507710011"
> <sip:4507710011 at 10.10.8.103;user=phone>;tag=89188638-a664e63f-ver5d
> From: <sip:811 at 10.10.10.205;user=phone>;tag=B2B.483.37
> CSeq: 2 INVITE
> Call-ID: 113c305d-89188638-113c3058-a664e63f at 10.10.8.103
> Max-Forwards: 70
> Content-Length: 0
> User-Agent: OpenSIPS (1.8.1-notls (x86_64/linux))
> Contact: <sip:10.10.10.205:5060;transport=tcp>
>
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 10.10.10.205;branch=z9hG4bK109d.ce759b97.0
> From: <sip:811 at 10.10.10.205;user=phone>;tag=B2B.483.37
> To: "4507710011"
> <sip:4507710011 at 10.10.8.103;user=phone>;tag=89188638-a664e63f-ver5d
> Call-ID: 113c305d-89188638-113c3058-a664e63f at 10.10.8.103
> CSeq: 2 INVITE
> Content-Length: 0
>
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.10.10.205;branch=z9hG4bK109d.ce759b97.0
> From: <sip:811 at 10.10.10.205;user=phone>;tag=B2B.483.37
> To: "4507710011"
> <sip:4507710011 at 10.10.8.103;user=phone>;tag=89188638-a664e63f-ver5d
> Call-ID: 113c305d-89188638-113c3058-a664e63f at 10.10.8.103
> CSeq: 2 INVITE
> Contact: <sip:4507710011 at 10.10.8.103:5060>
> Content-Type: application/sdp
> Content-Length: 185
>
> v=0
> o=VOIPSIL_SIP 416521046 416521046 IN IP4 10.10.8.103 s=Sip Call c=IN
> IP4 10.10.8.33
> t=0 0
> m=audio 21260 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
>
> ACK sip:4507710011 at 10.10.8.103;user=phone SIP/2.0
> Via: SIP/2.0/UDP 10.10.10.205;branch=z9hG4bK109d.de759b97.0
> To: "4507710011"
> <sip:4507710011 at 10.10.8.103;user=phone>;tag=89188638-a664e63f-ver5d
> From: <sip:811 at 10.10.10.205;user=phone>;tag=B2B.483.37
> CSeq: 2 ACK
> Call-ID: 113c305d-89188638-113c3058-a664e63f at 10.10.8.103
> Max-Forwards: 70
> Content-Length: 186
> User-Agent: OpenSIPS (1.8.1-notls (x86_64/linux))
> Allow: INVITE, ACK, CANCEL, BYE
> Priority: emergency
> Calluid: 1537A73F8B0C0000005C
> Geolocation: <cid:4507710011 at Sierra>
> Geolocation-Routing: yes
> Initial-CallID: 113c305d-89188638-113c3058-a664e63f at 10.10.8.103
> Contact: <sip:10.10.10.205:5060;transport=tcp>
> Content-Type: application/sdp
>
> v=0
> o=VOIPSIL_SIP 322321996 322321996 IN IP4 10.10.10.203 s=Sip Call c=IN
> IP4 10.10.10.7
> t=0 0
> m=audio 21336 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> BYE sip:4507710011 at 10.10.8.103;user=phone SIP/2.0
> Via: SIP/2.0/UDP 10.10.10.205;branch=z9hG4bK4037.cd227b95.0
> To: "4507710011"
> <sip:4507710011 at 10.10.8.103;user=phone>;tag=89188638-3ac4d92f-ver59
> From: <sip:811 at 10.10.10.205;user=phone>;tag=B2B.87.164
> CSeq: 3 BYE
> Call-ID: ae3179a8-89188638-ae3179a3-3ac4d92f at 10.10.8.103
> Max-Forwards: 70
> Content-Length: 0
> User-Agent: OpenSIPS (1.8.1-notls (x86_64/linux))
> Contact: <sip:10.10.10.205:5060;transport=tcp>
>
>
> OpenSIPS 1.11.5:
> ----------------
> NVITE sip:4507710011 at 10.10.8.103;user=phone SIP/2.0
> Via: SIP/2.0/TCP 10.10.10.205:5060;branch=z9hG4bK97f1.3d149882.0
> To: "4507710011"
> <sip:4507710011 at 10.10.8.103;user=phone>;tag=89188638-1dace611-ver5b
> From: <sip:811 at 10.10.10.205;user=phone>;tag=B2B.11.465
> CSeq: 2 INVITE
> Call-ID: 76579f54-89188638-76579f4f-1dace611 at 10.10.8.103
> Max-Forwards: 70
> Content-Length: 0
> User-Agent: OpenSIPS (1.11.5-notls (x86_64/linux))
> Contact: <sip:10.10.10.205:5060>
>
> SIP/2.0 100 Trying
> Via: SIP/2.0/TCP 10.10.10.205:5060;branch=z9hG4bK97f1.3d149882.0
> From: <sip:811 at 10.10.10.205;user=phone>;tag=B2B.11.465
> To: "4507710011"
> <sip:4507710011 at 10.10.8.103;user=phone>;tag=89188638-1dace611-ver5b
> Call-ID: 76579f54-89188638-76579f4f-1dace611 at 10.10.8.103
> CSeq: 2 INVITE
> Content-Length: 0
>
> SIP/2.0 200 OK
> Via: SIP/2.0/TCP 10.10.10.205:5060;branch=z9hG4bK97f1.3d149882.0
> From: <sip:811 at 10.10.10.205;user=phone>;tag=B2B.11.465
> To: "4507710011"
> <sip:4507710011 at 10.10.8.103;user=phone>;tag=89188638-1dace611-ver5b
> Call-ID: 76579f54-89188638-76579f4f-1dace611 at 10.10.8.103
> CSeq: 2 INVITE
> Contact: <sip:4507710011 at 10.10.8.103:5060>
> Content-Type: application/sdp
> Content-Length: 185
>
> v=0
> o=VOIPSIL_SIP 415919281 415919281 IN IP4 10.10.8.103 s=Sip Call c=IN
> IP4 10.10.8.33
> t=0 0
> m=audio 21260 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
>
> ACK sip:4507710011 at 10.10.8.103;user=phone SIP/2.0
> Via: SIP/2.0/TCP 10.10.10.205:5060;branch=z9hG4bK97f1.4d149882.0
> To: "4507710011"
> <sip:4507710011 at 10.10.8.103;user=phone>;tag=89188638-1dace611-ver5b
> From: <sip:811 at 10.10.10.205;user=phone>;tag=B2B.11.465
> CSeq: 2 ACK
> Call-ID: 76579f54-89188638-76579f4f-1dace611 at 10.10.8.103
> Max-Forwards: 70
> Content-Length: 0
> User-Agent: OpenSIPS (1.11.5-notls (x86_64/linux))
> Allow: INVITE, ACK, CANCEL, BYE
> Priority: emergency
> Calluid: 1537A68E4C0C0000005A
> Geolocation: <cid:4507710011 at Sierra>
> Geolocation-Routing: yes
> Initial-CallID: 76579f54-89188638-76579f4f-1dace611 at 10.10.8.103
> Contact: <sip:10.10.10.205:5060>
> Content-Type: application/sdp
>
> Louis Rochon
> Senior Systems Designer
> Solacom Technologies
>
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