Genesys CTI User Forum
Genesys CTI User Forum => Genesys CTI Technical Discussion => Topic started by: kirilpezev on January 31, 2011, 09:27:55 PM
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Hi,
I have a problem with a sip huge sip signaling overload.
When inbound sip call come from IP-to-IP Gateway to the SIP server and enter in routing strategy with several music treatments(music files). The SIP Server receives the Invite from the Gateway and replay with 200 OK. The call entered in routing strategy and plays the first announcement. Till now every think is ok. When the next music treatment in the routing strategy has to be played the following steps are happened:
1. For every treatment SIP Server send INVITE to Streaming Manager beginning with:
INVITE sip:annc@10.55.142.16:5090;play=announcement/1201;repeat=1 SIP/2.0
2. After that the Stream Manager replay to the SIP Server with 200 that in the body the provided RTP port that is change from the initial and is the next even:
m=audio 8872 RTP/AVP 8 101
3. In the next step SIP Server send new invite message to the Gateway(even the call is already establish) with the offered (in previous step) RTP port from the Stream Manager.
m=audio 8872 RTP/AVP 8 101
4. After that the Gateway replay again with the 200 ok and send INVITE to the Service Provider but not changed the RTP port, and use the initial one.
[b]All this steps are repeated for every separate music treatment in the URS routing strategy.[/b]
Because there are a lot of signaling messages (around 125 peaces only to play several music treatments /including 100 trying and ACK/) it results in very huge load on the Gateway.
[b]Is it possible on already established call SIP Server not to send intermediate INVITE messages to the Gateway for every music treatment? [/b]
Thanks in advance!!!
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Hi!
Please note that every time SIP Server has to re-direct the caller from one point to another, it has to send a re-INVITE. If you have a music treatment block with 2 or more audio files to play, then SIP Server will send a re-Invite to Stream Manager for each audio file to play, because the name of the audio file to play is included on INVITE message. When the customer is put on hold or retrieve from hold there will be a re-INVITE for each one of this action as well.
Bottom line: the fact that the call is established already won't prevent SIP Server to send as many re-INVITE as needed to provide the call flow as expected.
Regards,
Franklin.
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[quote author=kirilpezev link=topic=6145.msg26713#msg26713 date=1296509275]
Hi,
I have a problem with a sip huge sip signaling overload.
When inbound sip call come from IP-to-IP Gateway to the SIP server and enter in routing strategy with several music treatments(music files). The SIP Server receives the Invite from the Gateway and replay with 200 OK. The call entered in routing strategy and plays the first announcement. Till now every think is ok. When the next music treatment in the routing strategy has to be played the following steps are happened:
1. For every treatment SIP Server send INVITE to Streaming Manager beginning with:
INVITE sip:annc@10.55.142.16:5090;play=announcement/1201;repeat=1 SIP/2.0
2. After that the Stream Manager replay to the SIP Server with 200 that in the body the provided RTP port that is change from the initial and is the next even:
m=audio 8872 RTP/AVP 8 101
3. In the next step SIP Server send new invite message to the Gateway(even the call is already establish) with the offered (in previous step) RTP port from the Stream Manager.
m=audio 8872 RTP/AVP 8 101
4. After that the Gateway replay again with the 200 ok and send INVITE to the Service Provider but not changed the RTP port, and use the initial one.
[b]All this steps are repeated for every separate music treatment in the URS routing strategy.[/b]
Because there are a lot of signaling messages (around 125 peaces only to play several music treatments /including 100 trying and ACK/) it results in very huge load on the Gateway.
[b]Is it possible on already established call SIP Server not to send intermediate INVITE messages to the Gateway for every music treatment? [/b]
Thanks in advance!!!
[/quote]
No, it's not possible, this is how a SIP 2-way / 1pcc flow looks like.
How would the gateway know when to open a RTP stream otherwise?
Fra
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There is a way to optimize signaling but it requires upgrade to Media Server which will require upgrade to SIPS 8.0.4 and Framework at least 8.0. When Media Server is used there won't so many reINVITE's during queuing. Of course when call is transfered between agents or music on hold is played we need those reINVITE's
Pawel