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One-way audio after call is put on hold

unwell_11
Level 1
Level 1

 

Hi All,

Previously, we are using CUCM 9.1, H323 gateway, and ISDN for inbound and outbound calls without any issues.

Last night, we tried to use SIP for outbound calls to one of our local SIP providers. Everything seems to be working fine, but when testing the features, we encountered an issue with putting calls on hold.

Our set-up was:

CUCM --> sipv2 --> CUBE --> sipv2 --> ITSP

What happened is that if I make a call out from an SCCP phone (8941) to an external number, and then place that call on hold, the other party can hear music on hold fine. When I retrieve the call, they can hear me, but I cannot hear them.

I have these settings enabled on the SIP Profile.

- Early Offer support for voice and video calls (insert MTP if needed)

- Send send-receive SDP in mid-call INVITE

 

Looking at the CUBE SIP messages, here is a summary of what I saw:

- After I put the call on hold, i see re-INVITE from CUCM sent to CUBE with NO SDP (which is then forwarded to the ITSP).

- ITSP then responds with a 200OK with SDP that has a=sendrecv attribute which is received by CUBE

- CUBE then sends a 200 OK to CUCM with SDP, but without the a=sendrecv attribute.

- CUCM sends an ACK with a=sendonly to CUBE (which is then passed on to the ITSP). I believe this is fine, since at this stage the person on the other line could now hear MOH.

 

When I resume the call:

- another re-invite is sent with no SDP from CUCM to CUBE (which is forwarded on to the ITSP)

- ITSP sends to CUBE a 200 OK with SDP but, has a=recvonly attribute in it. This is in turn forwarded to CUCM, and CUCM responds with a=sendonly on its SDP.

 

I assumed that if you have Send send-receive SDP in mid-call INVITE ticked, it would do as it says - send an SDP during mid-call INVITEs. But based on the logs it's not doing what it's supposed to?

 

I tried ticking on "Media Termination Point Required" and this fixed the issue. However, reading some other posts here in the community, this is not the ideal fix.

 

If I could send a mid call INVITE with SDP, which will have the correct sendrecv attribute, it might probably fix the issue. But I can't seem to find any information to force Early Offer for mid-call invites?

 

Any assistance would be much appreciated. Thanks in advanced for the community's assistance!

18 Replies 18

This sorted my issue in the UK. Calls from Jersey Telecom into BT SIP 

erika.sasaki
Level 1
Level 1

old post but another way to resolve this without having to change the service parameter on CUCM is SIP profiles on the CUBE..for example,

 

voice class sip-profiles 1
 request ANY sdp-header Audio-Attribute modify "a=inactive" "a=sendrecv"
 request ANY sdp-header Audio-Attribute modify "a=recvonly" "a=sendrecv"

 

and then apply on dial-peers that are telco facing.

 

 

Thanks you Erika, you just solved a problem i had for 2 months in test lab using sipgate. Many thanks.

Erika you rock...!!!

 

I applied you solution to the router globally instead of to the dial-peer.  Thank you...!  You saved a headache...

 

 config t

 voice class sip-profiles 1
       request ANY sdp-header Audio-Attribute modify "a=inactive" "a=sendrecv"
       request ANY sdp-header Audio-Attribute modify "a=recvonly" "a=sendrecv"
      exit
      voice service voip
      sip
      sip-profiles 1
  end