Push video - SIP problem?

[alexandercoachman]

Hi,

when i want to push a video from Genesys Desktop, for example: announcement/101.zip, the SIP server wants to use an unknown ID to tell the Stream Manager to play the file. In the Genesys Desktop logs i dont find this ID, but in the  SIP server logs i found it:

14:37:48.187:(1) Sending  [420,UDP] 652 bytes to 10.35.37.10:5065 >>>>>
[b]INVITE sip:4e95849e00000007@10.35.37.10:5065 SIP/2.0[/b]
From: sip:1111@10.35.37.10:5060;tag=85FC5029-9FCC-4C4B-A481-782C352B6EE1-77
[b]To: <sip:4e95849e00000007@10.35.37.10:5065>[/b]
Call-ID: 925CDCD9-B2DC-4671-A58F-56C23E979278-32@10.35.37.10
CSeq: 1 INVITE
Content-Length: 0
Via: SIP/2.0/UDP 10.35.37.10:5060;branch=z9hG4bKECCD1C9D-2811-4208-8879-956AD38AD6B7-107
Contact: <sip:1111@10.35.37.10:5060>
Allow: ACK, BYE, CANCEL, INFO, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE
X-Genesys-Orig: agent
Max-Forwards: 69
X-Genesys-CallUUID: 3IP9LHNDK94012CT4R4QP7RC8G00000N
Session-Expires: 1800;refresher=uac
Min-SE: 90
Supported: timer


14:37:48.187:(1) SipDialog: event SEND_INVITE, t=778, s=2, r=6, m=030d6c44
14:37:48.187 SIPCONN(Multipoint Conference Unit): HandleSipDialogEvent(SEND_INVITE) - filtered
14:37:48.187 SIPCONN(Multipoint Conference Unit): sdp state SDP_STATE_NULL, event SDP_OFFER_REQUESTED
14:37:48.187 SIPCONN(Multipoint Conference Unit): new sdp state SDP_OFFER_REQUESTED, event SDP_OFFER_REQUESTED
14:37:48.187 --- CIFace::Request ---
14:37:48.187:(1) 14:37:48.187:(1) SIPS:LOGBLOCK:END:REQUEST:]
14:37:48.187:(1) SIPS:LOGBLOCK:BEGIN:SIPDATA:[
14:37:48.187:(1) Received [420,UDP] 399 bytes from 10.35.37.10:1190 <<<<<
[b]SIP/2.0 404 Not Found[/b]

But whats this ID? Which Genesys Component generates it and how should i configure my components to work correctly?

Thanks, Sandor

[cavagnaro]

404 means file not found...you sure about the path and configuration of your file?

[alexandercoachman]

Yes 404 is right, because i dont have announcement with this name: 4e95849e00000007
I would like to play a video file named: 101.zip

By the next call, this ID will be 4e95849e00000008. By the next: 4e95849e00000009, the next: 4e95849e0000000a ..and so on..

I dont know who generates this ID... and what it means.. 

[alexandercoachman]

Oooohhhh the problem was with my MCU configuration.
I added the option "prefix" with value "conf=" and now i have an another probleeem 

18:03:00.437 Trc 53055 SIPdialog[58] session info 11 (payload-changed) -- RTP(leg=584):8148
  codec 4(G.711/A-law) not acceptable (mask=u.....3.)

gsip:CL2CONN[280,UDP]:18:03:00.437 >>>> 437 bytes to 10.35.37.10:5060 >>>>
BYE sip:1112@10.35.37.10:5060 SIP/2.0
From: <sip:conf=4e95849e00000016@10.35.37.10:5065>;tag=6E34DF4C-8E14-4058-884A-C400FF82DFE3-57
To: sip:1112@10.35.37.10;tag=85FC5029-9FCC-4C4B-A481-782C352B6EE1-189
Call-ID: 925CDCD9-B2DC-4671-A58F-56C23E979278-110@10.35.37.10
CSeq: 1 BYE
Content-Length: 0
Via: SIP/2.0/UDP 10.35.37.10:5065;branch=z9hG4bK09B84306-A977-4F34-9CDA-7EE34678466A-3
Reason: SIP;cause=488;text=[b]"Bad RTP payload"[/b]

the call ends after i press the push video button. 

[René]

Hi Alex,

Please check Stream Manager's 'sip-conf-codecs' option in codecs section. Is G.711/A-law enabled?

R.

[alexandercoachman]

Okay, this option missed. Now another one. In the INVITE message i have a duplicated name for the announcement, SM will never find this file. I have a zipped video file named: 101.zip

I have a Video Service DN, with the following TServer option:

request-uri = annc@10.35.37.10:5065;play=announcement/101

SIPsetupLeg(audio)
RTPleg::setup [SM-7.6.004.01]-(rtp:8236)--(10.35.37.11:16744)
  (leg=844) state=e73[F<>.cirt] codec=3(G.711/mu-law) nte=101/97
SIPconf(id=4e95849e0000001d,audio) conferenced (rec=no)
  RTP(leg=844):8236-ssrc[59e0] Ag p-> 3(G.711/mu-law) in=0
  RTP(leg=834):8232-ssrc[2ff2] p-> 3(G.711/mu-law) in=13(0.24:0.42 sec)

SIPsetupLeg(video)
H263_config(cif4=2;cif=1;qcif=1;maxbr=10880)
RTPleg::setup [SM-7.6.004.01]-(rtp:8238)--(10.35.37.11:16746)
  (leg=845) state=e73[F<>.cirt] codec=15(H.263)
SIPconf(id=4e95849e0000001d,video) conferenced (rec=no)
  RTP(leg=845):8238-ssrc[76f5] Ag p-> 15(H.263) in=0
  RTP(leg=835):8234-ssrc[54ff] p-> 15(H.263) in=0/15


gsip:CL2LIST[304,UDP]:18:29:27.093 <<<< 719 bytes from 10.35.37.10:5060 <<<<
INVITE sip:conf=4e95849e0000001d@10.35.37.10:5065;confrole=push-all SIP/2.0
From: <sip:Video Service@10.35.37.10:5060>;tag=85FC5029-9FCC-4C4B-A481-782C352B6EE1-236
To: <sip:conf=4e95849e0000001d@10.35.37.10:5065;confrole=push-all>
Call-ID: 925CDCD9-B2DC-4671-A58F-56C23E979278-148@10.35.37.10
CSeq: 1 INVITE
Content-Length: 0
Via: SIP/2.0/UDP 10.35.37.10:5060;branch=z9hG4bKECCD1C9D-2811-4208-8879-956AD38AD6B7-1022
Contact: <sip:gcti::video@10.35.37.10:5060>
Allow: ACK, BYE, CANCEL, INFO, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE
X-Genesys-Orig: agent
Max-Forwards: 69
X-Genesys-CallUUID: 3IP9LHNDK94012CT4R4QP7RC8G00001U
Session-Expires: 1800;refresher=uac
Min-SE: 90
Supported: timer


SIPdialog[85] event 12 INVITE
18:29:27.093 Trc 53050 SIPdialog[85] conf=4e95849e0000001d;confrole=push-all
  RTPLeg[854] created RTP:8240(fd=392), RTCP:8241(fd=332)
  RTPLeg[855] created RTP:8242(fd=336), RTCP:8243(fd=352)
OfferSDP(0|ua39s.34)

gsip:CL2CONN[280,UDP]:18:29:27.093 >>>> 896 bytes to 10.35.37.10:5060 >>>>
SIP/2.0 200 OK
From: <sip:Video Service@10.35.37.10:5060>;tag=85FC5029-9FCC-4C4B-A481-782C352B6EE1-236
To: <sip:conf=4e95849e0000001d@10.35.37.10:5065;confrole=push-all>;tag=6E34DF4C-8E14-4058-884A-C400FF82DFE3-84
Call-ID: 925CDCD9-B2DC-4671-A58F-56C23E979278-148@10.35.37.10
CSeq: 1 INVITE
Via: SIP/2.0/UDP 10.35.37.10:5060;branch=z9hG4bKECCD1C9D-2811-4208-8879-956AD38AD6B7-1022;received=10.35.37.10
Contact: <sip:10.35.37.10:5065>
Content-Type: application/sdp
Content-Length: 401

v=0
o=Genesys 113 113 IN IP4 10.35.37.10
s=StreamManager 7.6.004.01 conf
c=IN IP4 10.35.37.10
t=0 0
m=audio 8240 RTP/AVP 0 8 3 4 18 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:4 g723/8000
a=rtpmap:18 g729/8000
m=video 8242 RTP/AVP 34 108
a=rtpmap:34 h263/90000
a=rtpmap:108 h264/90000
a=x-media-op:conf

SIPdialog[85] event 14 CALLED/ResOK

gsip:CL2LIST[304,UDP]:18:29:27.093 <<<< 1170 bytes from 10.35.37.10:5060 <<<<
[b]INVITE sip:annc@10.35.37.10:5065;play=announcement/101101 SIP/2.0[/b]
From: <sip:conf=4e95849e0000001d@10.35.37.10:5065;confrole=push-all>;tag=85FC5029-9FCC-4C4B-A481-782C352B6EE1-237
To: <sip:Video Service@10.35.37.10:5060>
Call-ID: 925CDCD9-B2DC-4671-A58F-56C23E979278-149@10.35.37.10
CSeq: 1 INVITE
Content-Length: 406
Content-Type: application/sdp
Via: SIP/2.0/UDP 10.35.37.10:5060;branch=z9hG4bKECCD1C9D-2811-4208-8879-956AD38AD6B7-1023
Contact: <sip:Multipoint Conference Unit@10.35.37.10:5060>
Allow: ACK, BYE, CANCEL, INFO, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE
X-Genesys-Orig: agent
Max-Forwards: 69
X-Genesys-CallUUID: 3IP9LHNDK94012CT4R4QP7RC8G00001U
Session-Expires: 1800;refresher=uac
Min-SE: 90
Supported: 100rel,timer

v=0
o=Genesys 1318416568 1 IN IP4 10.35.37.10
s=StreamManager 7.6.004.01 conf
c=IN IP4 10.35.37.10
t=0 0
m=audio 8240 RTP/AVP 0 8 3 4 18 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:4 g723/8000
a=rtpmap:18 g729/8000
m=video 8242 RTP/AVP 34 108
a=x-media-op:conf
a=rtpmap:34 h263/90000
a=rtpmap:108 h264/90000

SIPdialog[86] event 12 INVITE
18:29:27.093 Trc 53050 SIPdialog[86] annc;play=announcement/101101
  m=audio RTPport=8240
    pt=0 codec=3(G.711/mu-law)
    pt=8 codec=4(G.711/A-law)
    pt=3 codec=10(GSM)
    pt=4 codec=5(G.723.1)
    pt=18 codec=7(G.729a)
    pt=101 telephone-event
  m=video RTPport=8242
    pt=34 codec=15(H.263)
    pt=108 codec=16(H.264)
Probing announcement/101101 (codecs=ua39s.34,fmt=wav)
  announcement/101101_{pcmu,mulaw}.wav not found
  announcement/101101_{pcma,alaw}.wav not found
  announcement/101101_{g723,g7231}.wav not found
  announcement/101101_{g729,g729a}.wav not found
  announcement/101101_{gsm,gsmFR}.wav not found
  announcement/101101_h263_{CIF4=1,CIF4=2,CIF=1,CIF=2,QCIF=1,QCIF=2} not found
  announcement/101101_h264_{42e01e=1,42e016=1,2e014=1,42e00d=1,42e00d=2,42e01c=1} not found
Probing announcement/101101 (codecs=....m...,fmt=wav)
  announcement/101101_{msgsm,gsmF}.wav not found

gsip:CL2CONN[280,UDP]:18:29:27.093 >>>> 436 bytes to 10.35.37.10:5060 >>>>
SIP/2.0 404 Not Found

I could be soo soo happy when it could work... 

[cavagnaro]

The issue is this:
[quote]
Probing announcement/101101 (codecs=ua39s.34,fmt=wav)
  announcement/101101_{pcmu,mulaw}.wav not found
  announcement/101101_{pcma,alaw}.wav not found
  announcement/101101_{g723,g7231}.wav not found
  announcement/101101_{g729,g729a}.wav not found
  announcement/101101_{gsm,gsmFR}.wav not found
  announcement/101101_h263_{CIF4=1,CIF4=2,CIF=1,CIF=2,QCIF=1,QCIF=2} not found
  announcement/101101_h264_{42e01e=1,42e016=1,2e014=1,42e00d=1,42e00d=2,42e01c=1} not found
Probing announcement/101101 (codecs=....m...,fmt=wav)
  announcement/101101_{msgsm,gsmF}.wav not found
[/quote]

You have to follow that naming convention

[alexandercoachman]

Yes i know. I can play videofiles to clients, like Tandberg E20, or MXP1700 hardvare videoconferencing systems.

But, i have a videofile named: 101.zip. I dont know why it appears in the logs as a duplicated name. But when i rename it to 101101, its better but not good. When i push the "push video" button, on the agents tandberg E20 i have a lot of codec problems in the live stream.. On the customer side i can hear only the voice of the video.. its better, but...

Thanks for the help, but i give it up for now.

[cavagnaro]

Can you post a screenshot of your strategy detailing treatment object config?

[alexandercoachman]

I'm at home now, but in the strategy i use a simple "play announcement" block, with a simple voice prompt pointed to the video file. When i call this RP, it works fine.
Sorry, but next days i travel to an another city to implement our demo videocc server. We will demonstrate how it works on the Budapest Calling expo. Budapest Calling is an independent expo of products and services related to call center profession in Hungary.

We have a lot of problems with our clients, because we're using Tandberg video hardvares - unsupported all of them by Genesys. We are using the Requestec's Flash2SIP gateway solution, Tandberg Movi integration, but we have a lot of problems with the videos..  Tandberg EX90 doesnt play the video, it crashes, but it can call an agent through Genesys routing. Tandberg E20 is great SIP client, it is the only client that works perfectly. Tandberg MXP1700 cant play CIF4 videos in good quality.. Video on hold, and Push video features dont work..

Video on hold is almost good, but when it plays the video to the customer, sometime they can talk with the agent, beause they dont hear the voice of the video. But sometimes they can not talk, because they can hear the voice of the video..  ... AND when I push the Hold button again, i cant retrieve the customer call.. it plays a video from the start. Its interesting, here at home in my private genesys 8 system i can push the button to retrieve the call  We dont have more time to play with it...

But, this is a demo system.. and we are new by Genesys.. And i think Genesys is very hard, there is a lot of options, applications and the deployment guides are detailed insufficiently for the newbies.

Best Regards
Sandor