Hello Sunil,

I checked it the offer/answer of this call and unable to find anything like
that....

The only thing i have noticed which uttam has mentioned that the SSRC are
different for both the RTP packets which is going from SBC and the one which
is coming to SBC...

Below is the SDP information in UDP format.... could you please  help me out
on this.... or i am not sure why it is giving the two SSRC value in same
dialog...

INVITE 
sip:[email protected]<sip%[email protected]>SIP/2.0
Max-Forwards: 69
Session-Expires: 3600;refresher=uac
Supported: timer, 100rel
To: <sip:[email protected]<sip%[email protected]>
>
From: <sip:8003223...@tmsash1-15-sip>;tag=3468945569-250042
Call-ID: [email protected]
CSeq: 1 INVITE
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER,
SUBSCRIBE, PRACK, UPDATE
Via: SIP/2.0/UDP 8.15.244.10:5060
;branch=z9hG4bKf5de4d0e3c0860d122139e1e30a2995c
Contact: <sip:[email protected]:5060>
Call-Info:
<sip:8.15.244.10>;method="NOTIFY;Event=telephone-event;Duration=1000"
Content-Type: application/sdp
Content-Length: 124

v=0
o=mscasheq01 547500705 547500705 IN IP4 8.15.244.10
s=sip call
c=IN IP4 8.15.244.11
t=0 0
m=audio 19612 RTP/AVP 0

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 8.15.244.10:5060
;branch=z9hG4bKf5de4d0e3c0860d122139e1e30a2995c
From: <sip:8003223...@tmsash1-15-sip>;tag=3468945569-250042
To: <sip:[email protected]<sip%[email protected]>
>
Call-ID: [email protected]
CSeq: 1 INVITE
Server: Sansay VSX 2.1
Content-Length: 0

SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 8.15.244.10:5060
;branch=z9hG4bKf5de4d0e3c0860d122139e1e30a2995c
From: <sip:8003223...@tmsash1-15-sip>;tag=3468945569-250042
To: <sip:[email protected]<sip%[email protected]>
>;tag=228209687-tdb1145808424
Call-ID: [email protected]
CSeq: 1 INVITE
Server: Sansay VSX 2.1
Content-Length: 0

SIP/2.0 200 OK
Via: SIP/2.0/UDP 8.15.244.10:5060
;branch=z9hG4bKf5de4d0e3c0860d122139e1e30a2995c
From: <sip:8003223...@tmsash1-15-sip>;tag=3468945569-250042
To: <sip:[email protected]<sip%[email protected]>
>;tag=228209687-tdb1145808424
Call-ID: [email protected]
CSeq: 1 INVITE

Server: Sansay VSX 2.1
Contact: <sip:[email protected]:5060;transport=udp>
Content-Type: application/sdp
Content-Length:   158

v=0
o=sansay-VSX 10 10 IN IP4 64.156.174.71
s=session controller
c=IN IP4 64.156.174.71
t=0 0
m=audio 10766 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20

ACK sip:[email protected]:5060;transport=udp SIP/2.0
Max-Forwards: 69
To: <sip:[email protected]<sip%[email protected]>
>;tag=228209687-tdb1145808424
From: <sip:8003223...@tmsash1-15-sip>;tag=3468945569-250042
Call-ID: [email protected]
CSeq: 1 ACK
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER,
SUBSCRIBE, PRACK, UPDATE
Via: SIP/2.0/UDP 8.15.244.10:5060
;branch=z9hG4bK7208e1bfdd2962741dc3dee9d2576820
Contact: <sip:[email protected]:5060>
Content-Length: 0

Thanks,
Nitin Kapoor





2009/12/30 <[email protected]>

> Hi Nitin,
>
> I think you need to check the SDP information in offer and answer and
> link the connection IP and port along with what you have offered and
> received in answer. Check the RTP originating and terminating IP and
> port. I think for some reason you are sending the same connection IP and
> port in two different offers on different call legs.
>
> Regards
> Sunil Verma
>
> -----Original Message-----
> From: [email protected]
> [mailto:[email protected]] On Behalf Of
> Nitin Kapoor
> Sent: Tuesday, December 29, 2009 11:18 PM
> To: [email protected]
> Subject: [Sip-implementors] Two Media Stream in one Dialog
>
> Hello All,
>
> I am facing the issue with one of my customer where he is getting two
> media
> streams in one call and they are using SBC(Nextone)
>
> I checked the signaling & RTP traces and everything looks okay to me.
>
> As per the traces one  RTP packet is going out and the other is coming
> in, *But
> whenever i am decoding the packets and trying to hear the conversation
> so
> both are the different conversation. *
>
> Could you please let me know is it possible where i can have the two
> media
> streams in one call when this is the not the conference call. And if
> this is
> possible then what is the scenario where its possible.
>
> Thanks,
> Nitin Kapoor
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