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One way voice when answered from hunt queue

fei he
Explorer
Explorer

Hi

We recently cross an issue where one way audio when answer from hunt queued calls and call flow as below:

ITSP ---sip---> CUBE(SP) --- sip--> CUCM --- sip---> ip phone

PSTN caller 1 call the hunt and answer by IP phone with 2 way audio. Then 2nd PSTN call coming into hunt which configured for Native queuing and 2nd call is queued as all hunt members are already on the call and PSTN caller 2 hear music on hold. IP phone disconnect PSTN caller 1 and PSTN caller 2 previous queued  is ringing to IP Phone and PSTN caller 2 start hear ring back tone. IP Phone answer the call and one-way audio occurs where PSTN caller 2 can hear IP Phone but IP Phone cannot hear PSTN caller 2 (same call flow works perfectly fine when all calling and called party from internal ip phone).

Has anyone came cross similar issue and any suggestion?

Fei

4 Replies 4

Dennis Mink
Advisor
Advisor

Sound like either a:  a routing issue (ping between phones and PSTN gateway).

or b, which is most likely, a signalling issue.  when your caller called into the native queueing then the queue will send to your caller that RTP=inactive and it will only play out the music on hold. I am guessing that when a phone becomes available, there is no re-invite to signal that RTP is now 2ways.

can you trace the call and send us the SIP traces.

cheers

Please remember to rate useful posts, by clicking on the stars below.

2nd call coming to the Hunt and call is been queued (on hold), call flow shows Carriage send the original INVITE to the CUBE, CUBE replies and sends a 200OK/SDP for answer with media attribute sendonly.
 

00171775.002 |17:08:09.600 |AppInfo  |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.111.3.253 on port 51977 index 2571831 with 1033 bytes:[525958909,NET]
INVITE sip:+61262644589@10.111.2.5:5060 SIP/2.0
Via: SIP/2.0/TCP 10.111.3.253:5060;branch=z9hG4bK+089560f63f02643084aba99e578d54281+sip+1+ac27ca87
Supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join,x-nortel-sipvc,gin,com.nortelnetworks.im.encryption,tdialog
From:  <sip:+61280854712@10.111.3.253:5060>;tag=10.111.3.253+1+188e73cd+54b8b409
To: <sip:+61262644589@10.111.2.5:5060>
CSeq: 3996 INVITE
Expires: 180
Content-Length: 190
Call-Info: <sip:10.111.3.253:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Contact: <sip:10.111.3.253:5060;transport=tcp>
Content-Type: application/sdp
Call-ID: 0e6aafbd5f893e9aadc1978a16c0a3c5@10.111.3.253
Max-Forwards: 49
Allow: INVITE,BYE,CANCEL,ACK,REGISTER,SUBSCRIBE,NOTIFY,MESSAGE,INFO,REFER,OPTIONS,PUBLISH,PRACK
Accept: application/sdp, application/dtmf-relay
v=0
o=PVG 65156409607295 65156409607295 IN IP4 10.111.3.254
s=-
c=IN IP4 10.111.3.254
t=0 0
m=audio 27190 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
 
 
00172133.001 |17:08:09.788 |AppInfo  |SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.111.3.253 on port 51977 index 2571831 [525958974,NET]
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.111.3.253:5060;branch=z9hG4bK+089560f63f02643084aba99e578d54281+sip+1+ac27ca87
From: <sip:+61280854712@10.111.3.253:5060>;tag=10.111.3.253+1+188e73cd+54b8b409
To: <sip:+61262644589@10.111.2.5:5060>;tag=1590641452~8f65986a-1de3-4cbe-a6b4-16bb52492403-43429771
Date: Tue, 07 Mar 2017 06:08:09 GMT
Call-ID: 0e6aafbd5f893e9aadc1978a16c0a3c5@10.111.3.253
CSeq: 3996 INVITE
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
Allow-Events: presence
Supported: replaces
Server: Cisco-CUCM10.5
Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=MIXED
Supported: X-cisco-srtp-fallback
Supported: Geolocation
P-Asserted-Identity: <sip:+61262644589@10.111.2.5>
Remote-Party-ID: <sip:+61262644589@10.111.2.5>;party=called;screen=yes;privacy=off
Contact: <sip:+61262644589@10.111.2.5:5060;transport=tcp>
Content-Type: application/sdp
Content-Length: 225
v=0
o=CiscoSystemsCCM-SIP 1590641452 1 IN IP4 10.111.2.5
s=SIP Call
c=IN IP4 10.111.3.4
t=0 0
m=audio 4000 RTP/AVP 8 101
a=ptime:20
a=rtpmap:8 PCMA/8000
a=sendonly                  --------------> CUBE instruct EVOLVE this call been queued or on hold.
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15


Once the queued call is answered, there is a re-INVITE coming from the CUBE and the Carriage replies with 200OK/SDP recvonly which is causing the one way speech.

00185286.001 |17:08:21.728 |AppInfo  |SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.111.3.253 on port 5060 index 2401885 [525962245,NET]
INVITE sip:10.111.3.253:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.111.2.5:5060;branch=z9hG4bK2bec81347f5a2cd
From: <sip:+61262644589@10.111.2.5:5060>;tag=1590641452~8f65986a-1de3-4cbe-a6b4-16bb52492403-43429771
To: <sip:+61280854712@10.111.3.253:5060>;tag=10.111.3.253+1+188e73cd+54b8b409
Date: Tue, 07 Mar 2017 06:08:21 GMT
Call-ID: 0e6aafbd5f893e9aadc1978a16c0a3c5@10.111.3.253
Supported: timer,resource-priority,replaces
User-Agent: Cisco-CUCM10.5
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 104 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=DESKTOP
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Min-SE:  1800
P-Asserted-Identity: <sip:+61262643770@10.111.2.5>
Remote-Party-ID: <sip:+61262643770@10.111.2.5>;party=calling;screen=yes;privacy=off
Contact: <sip:367e1c51-af86-2224-e512-936a09af226d@10.111.2.5:5060;transport=tcp>
Content-Type: application/sdp
Content-Length: 229
v=0
o=CiscoSystemsCCM-SIP 1590641452 4 IN IP4 10.111.2.5
s=SIP Call
c=IN IP4 10.136.156.120
b=TIAS:64000
b=AS:64
t=0 0
m=audio 18254 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
 
 
00185341.002 |17:08:21.774 |AppInfo  |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.111.3.253 on port 5060 index 2401885 with 803 bytes:
[525962266,NET]
SIP/2.0 200 OK
Supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join,x-nortel-sipvc,gin,com.nortelnetworks.im.encryption
Call-ID: 0e6aafbd5f893e9aadc1978a16c0a3c5@10.111.3.253
CSeq: 104 INVITE
From: <sip:+61262644589@10.111.2.5:5060>;tag=1590641452~8f65986a-1de3-4cbe-a6b4-16bb52492403-43429771
To: <sip:+61280854712@10.111.3.253:5060>;tag=10.111.3.253+1+188e73cd+54b8b409
Via: SIP/2.0/TCP 10.111.2.5:5060;received=10.111.2.5;branch=z9hG4bK2bec81347f5a2cd
Content-Length: 224
Contact: <sip:10.111.3.253:5060;transport=tcp>
Content-Type: application/sdp
v=0
o=PVG 65156409607295 65156409607296 IN IP4 10.111.3.254
s=-
c=IN IP4 10.111.3.254
t=0 0
m=audio 27190 RTP/AVP 8 101
a=recvonly                 --------------> this cause one way speech
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

This issue is now resolved after we uncheck "Send send-receive SDP in mid-call INVITE" on SIP profile used by trunk.

By looking at the traces, the invite from the second call comes with no codec. The 200 OK comes with a codec and a send only. so when the 2nd call gets picked up, there is a codec towards PSTN (g711alaw), but no codec i.e. voice from PSTN.

I would reach out to your provider, and ask them why that Eoraly Offer SDP does not contain any codecs.

Please rate if this helps.

Please remember to rate useful posts, by clicking on the stars below.

Hello I have the same problems. I have try all options.

at the end the log files.

thanks!

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