Lync new INVITE behavior when calling (invite by phone) PSTN Numbers into a Conference

  • Article

It has been a while since we last posted something new to our blog, but we are back. 

I wanted to start a new tread regarding new or changed features and behavior in Lync 2010 as opposed to Office Communications Server 2007 R2.

Based on our customers feedback the behavior has been changed when inviting a PSTN Number from a Lync meeting, the change of behavior might need some configuration of your Gateway/PBX if you are migrating your Mediation Server from Office Communications Server 2007 R2 to Lync.

We have seen numerous issues with billing in Office Communications Server 2007 R2 when dialing out from a meeting, this is because in R2 when you were dialing out from a meeting, the call was initiated by the meeting service and not by the user itself, the result of this was that a generic INVITE Message was being sent to the Gateway with an anonymous caller ID and no way to identify the User actually making the call.

Some customers that were using a SIP Trunk and where the billing was done based on DID could so not make any outbound calls from the Meeting as the SIP TRUNK/Gateway was refusing Anonymous Calls.

In Lync we now have a new behavior, the call is still made by the Conferencing Service but this time the Conferencing Service got smarter, it no longer sends a plain anonymous Invite to the Gateway but identifies the user making the call and uses this users Number to make the call.

Here are a few examples of the behavior.

  • SIP INVITE to Gateway in R2 when a PSTN User was invited to an existing meeting

 

22d:22h:57m:22s INVITE sip:+402681230001@10.165.213.38;user=phone SIP/2.0

FROM: <sip:testuser@vonfluxenberg.tk;gruu;opaque=app:conf:audio-video:id:06D549EC9F7EFD4F963819F7B7DAE248>;epid=8327676970;tag=6811c6f52b

TO: <sip:+402681230001@10.165.213.38;user=phone>

CSEQ: 12 INVITE

CALL-ID: 42c9c8d7-3390-4e01-a01f-51c70c368f71

MAX-FORWARDS: 70

VIA: SIP/2.0/TCP 192.168.0.4:59804;branch=z9hG4bK45caa86

CONTACT: <sip:VFMS.vonfluxenberg.tk:5060;transport=Tcp;maddr=192.168.0.4;ms-opaque=a7c3a97fdf3f6c79>

CONTENT-LENGTH: 324

SUPPORTED: 100rel

USER-AGENT: RTCC/3.5.0.0 MediationServer

CONTENT-TYPE: application/sdp; charset=utf-8

ALLOW: ACK

Allow: CANCEL,BYE,INVITE,PRACK,UPDATE

v=0

o=- 18 1 IN IP4 192.168.0.4

s=session

c=IN IP4 192.168.0.4

b=CT:1000

t=0 0

m=audio 60152 RTP/AVP 97 101 13 0 8

c=IN IP4 192.168.0.4

a=rtcp:60153

a=label:Audio

a=rtpmap:97 RED/8000

a=rtpmap:101 telephone-event/8000

Notice how the FROM Field (FROM: <sip:testuser@vonfluxenberg.tk;gruu;opaque=app:conf:audio-video:id:06D549EC9F7EFD4F963819F7B7DAE248>;epid=8327676970;tag=6811c6f52b) Contains no E164 Number, just a Username of the User that initiated the meeting and the conf tag, this is displayed here because the conferencing service initiated the call.

As you can imagine, most PBX/Gateways will have no idea who or what testuser@vonfluxenberg.tkis as they are designed to work with E164 Numbers and not SIP Addresses. At best they would treat this as an Anonymous call but PBX’s or SIP TRUNKS that rely on DID for Billing would just simply decline the call…. Not very fun for some.

 

  • SIP INVITE to Gateway in Lync 2010 when a PSTN User was invited to an existing meeting

 

22d:22h:55m:2s INVITE sip:+402681230001@10.165.213.38;user=phone SIP/2.0

FROM: <sip:+49891242666@vflync.vonfluxenberg.tk;user=phone>;epid=79D93FDA4B;tag=5b1fb1ee8

TO: <sip:+402681230001@10.165.213.38;user=phone>

CSEQ: 2392 INVITE

CALL-ID: 9fcf1195-2ec7-4039-a4e6-1ba12bbb8c7c

MAX-FORWARDS: 70

VIA: SIP/2.0/TCP 192.168.2.2:56229;branch=z9hG4bK336c7520

CONTACT: <sip:vflync.vonfluxenberg.tk:5068;transport=Tcp;maddr=192.168.2.2;ms-opaque=468b0470ce4bcf63>

CONTENT-LENGTH: 335

SUPPORTED: 100rel

USER-AGENT: RTCC/4.0.0.0 MediationServer

CONTENT-TYPE: application/sdp

ALLOW: ACK

Allow: CANCEL,BYE,INVITE,PRACK,UPDATE

v=0

o=- 4 1 IN IP4 192.168.2.2

s=session

c=IN IP4 192.168.2.2

b=CT:1000

t=0 0

m=audio 52548 RTP/AVP 97 101 13 0 8

c=IN IP4 192.168.2.2

a=rtcp:52549

a=label:Audio

a=sendrecv

a=rtpmap:97 RED/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=rtpmap:13 CN/8000

a=rtpmap

                As opposed to the example before, notice how the SIP Invite from the Lync 2010 Server contains in the From Field (FROM:sip:+49891242666@vflync.vonfluxenberg.tk;user=phone>;epid=79D93FDA4B;tag=5b1fb1ee8) a telephone number and no longer the SIP User Name as in R2, this will make SIP TRUNK Providers that rely on DID’s for billing very happy as they can now charge you for the Conference Calls but it can help in other scenarios also.

Hope you have found this information helpful and we promise to be back soon with more tips and tricks on LYNC and OCS.