Two M line handing for SIPREC recording.

[rahulsonawane1518]

We are trying to make freeswitch as call recording server for which I am using SIPREC on my media gateway.

When gateway is sending an Invite to FS, in SDP it sends 2 m line. One m line for customer and another m line for internal.

However when FS is responding 200 OK, It is only selecting first m line, beacuse of which only one leg call getting recorded.

IS there any way, when FS will respond, It response with two m lines.

INVITE from media Gateway

INVITE sip:7021@1.0.0.4;user=phone SIP/2.0
Via: SIP/2.0/UDP 1.0.0.3:5060;branch=z9hG4bKac567880035
Max-Forwards: 70
From: sip:1.0.0.3;user=phone;tag=1c1611607574
To: sip:7021@1.0.0.4
Call-ID: 3989760942622025111012@1.0.0.3
CSeq: 1 INVITE
Contact: sip:1.0.0.3:5060;+sip.src
Supported: replaces,resource-priority,sdp-anat
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Mediant VE SBC/v.7.40A.251.150
Content-Type: multipart/mixed;boundary=boundary_ac173e
Content-Length: 2404

--boundary_ac173e
Content-Type: application/sdp

v=0
o=AudiocodesGW 964480698 33218309 IN IP4 1.0.0.3
s=SBC-Call
c=IN IP4 1.0.0.3
t=0 0
m=audio 19980 RTP/AVP 0 101
c=IN IP4 1.0.0.3
a=ptime:20
a=sendonly
a=label:1
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15,16
m=audio 19984 RTP/AVP 0 101
c=IN IP4 1.0.0.3
a=ptime:20
a=sendonly
a=label:2
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15,16

---

200 OK from FS

SIP/2.0 200 OK
Via: SIP/2.0/UDP 1.0.0.3:5060;branch=z9hG4bKac567880035
From: sip:1.0.0.3;user=phone;tag=1c1611607574
To: sip:7021@1.0.0.4;tag=55t2Fmp88vFjg
Call-ID: 3989760942622025111012@1.0.0.3
CSeq: 1 INVITE
Contact: sip:7021@1.0.0.6:5060;transport=udp
User-Agent: Jio
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 244
Remote-Party-ID: "7021" sip:7021@1.0.0.4;party=calling;privacy=off;screen=no

v=0
o=root 1740518086 1740518087 IN IP4 1.0.0.6
s=root
c=IN IP4 1.0.0.6
t=0 0
m=audio 29318 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=recvonly
a=ptime:20
m=audio 0 RTP/AVP 19