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SIP INFO during INVITE

SOLVED

SIP INFO during INVITE

Hey there,

 

We are currently looking at trying to provide a solution within our application to allow desk phones to answer a call whilst it’s ringing.

 

We have looked at different methods and found that we could provide SIP INFO containing the auto answer header however the Polycom does not accept this.

 

Do you know of any method of getting the Polycom to answer via a SIP packet mid INVITE

 

I have attached a sample PCAP of what we would like to achieve

 

Kind Regards

Warren

1 ACCEPTED SOLUTION

Accepted Solutions
SteffenBaierUK
Polycom Employee & Community Manager

Re: SIP INFO during INVITE

Hello Warren,

I followed this up with one of our most senior and experienced engineers and got the following response.

 

  • In our 180 responses we list that we accept the "talk" event.  A NOTIFY on the same dialog (ie: matching to/from tags & call-id) with that event will trigger an immediate answer

Polycom 180 reply:

 

0504201613|sip  |0|00|>>> Data Send to 10.252.122.122:5060
0504201613|sip  |0|00|    SIP/2.0 180 Ringing
0504201613|sip  |0|00|    Via: SIP/2.0/UDP 10.252.122.122:5060;branch=z9hG4bK7b03f142
0504201613|sip  |0|00|    From: "Steffen 1" <sip:3080@10.252.122.122>;tag=as2ab9a3ef
0504201613|sip  |0|00|    To: "3070" <sip:3070@10.252.149.107>;tag=245EF5BF-90A06E1C
0504201613|sip  |0|00|    CSeq: 102 INVITE
0504201613|sip  |0|00|    Call-ID: 5ea2e0436ea5963a1717b93f57548480@10.252.122.122:5060
0504201613|sip  |0|00|    Contact: <sip:3070@10.252.149.107>
0504201613|sip  |0|00|    User-Agent: Polycom/5.4.1.14510 PolycomVVX-VVX_500-UA/5.4.1.14510
0504201613|sip  |0|00|    Allow-Events: conference,talk,hold
0504201613|sip  |0|00|    Accept-Language: en
0504201613|sip  |0|00|    Content-Length: 0
0504201613|sip  |0|00|    
0504201613|sip  |0|00|<<< End of data send

 

Basically sending a SIP Info

 

SIP_INFO_Event_Talk.png

 

With the Event:talk would auto answer the call.

Please ensure to provide some feedback if this reply has helped you so other users can profit from your experience.

Best Regards

Steffen Baier

Polycom Global Services

----------------
The title Polycom Employee & Community Manager is a community setting and does not reflect my role. I am just a simple volunteer in the community like everybody else. My official "day" Job is 3rd Level support at Poly but I am unable to provide official support via the community.

----------------

Notice: This community forum is not an official Poly support resource, thus responses from Poly employees, partners, and customers alike are best-effort in attempts to share learned knowledge. If you need immediate and/or official assistance please open a service ticket through your proper support channels.
Please also ensure you always check the VoIP , Video Endpoint , Skype for Business , PSTN or RPM FAQ's

View solution in original post

Message 7 of 8
7 REPLIES 7
SteffenBaierUK
Polycom Employee & Community Manager

Re: SIP INFO during INVITE

Hello Warren,

welcome to the Polycom Community.

The community's VoIP FAQ contains this post here:

Oct 25, 2011 Question: How can I change my Ringtone or Ring in a special manner for a certain incoming call?

Resolution: Please check => here <=

 

Looking at your wireshark you are using the "wrong" format. The above FAQ post has all the details as we look for a Alert-Info field.

Please ensure to provide some feedback if this reply has helped you so other users can profit from your experience.

Best Regards

Steffen Baier

Polycom Global Services

----------------
The title Polycom Employee & Community Manager is a community setting and does not reflect my role. I am just a simple volunteer in the community like everybody else. My official "day" Job is 3rd Level support at Poly but I am unable to provide official support via the community.

----------------

Notice: This community forum is not an official Poly support resource, thus responses from Poly employees, partners, and customers alike are best-effort in attempts to share learned knowledge. If you need immediate and/or official assistance please open a service ticket through your proper support channels.
Please also ensure you always check the VoIP , Video Endpoint , Skype for Business , PSTN or RPM FAQ's
Message 2 of 8

Re: SIP INFO during INVITE

Thank you for your response Steffen, sorry I provided the wrong wireshark trace as this was intended for our softphone.

 

We do provide the Call-info header (Call-Info: <sip:>;answer-after=0) on the INVITE which works perfectly fine, however my question here is how can I get the phone to auto answer but after we have sent the initial INVITE and before the polycom has responded with a 200.

 

The idea is:

 

Our application sends a standard invite without answer-after

Phone is ringing

Customer presses ANSWER in our application

We send an SIP INFO packet with (Call-Info: <sip:>;answer-after=0) in the header

Polycom will then answer that call

 

Kind Regards

Warren

 

Message 3 of 8
SteffenBaierUK
Polycom Employee & Community Manager

Re: SIP INFO during INVITE

Hello Warren,

It would be usefull if you provided additional information on what actual phones you want to use this as the answer depends on this.

 

To my knowledge we only look at an INVITE for the call header and not a SIP INFO.

 

You could try this:

 

Feb 04, 2013 Question: Can I remotely control the Phone or send content to the Phone?

Answer: The <apps/> parameter can be used to utilize the push server controls and more information can be found => here <=

 

Apr 21, 2015 Question: Can I use advanced remote control possibilities / CTI?

Resolution: Please check => here <=


Please ensure to provide some feedback if this reply has helped you so other users can profit from your experience.

Best Regards

Steffen Baier

Polycom Global Services

----------------
The title Polycom Employee & Community Manager is a community setting and does not reflect my role. I am just a simple volunteer in the community like everybody else. My official "day" Job is 3rd Level support at Poly but I am unable to provide official support via the community.

----------------

Notice: This community forum is not an official Poly support resource, thus responses from Poly employees, partners, and customers alike are best-effort in attempts to share learned knowledge. If you need immediate and/or official assistance please open a service ticket through your proper support channels.
Please also ensure you always check the VoIP , Video Endpoint , Skype for Business , PSTN or RPM FAQ's
Message 4 of 8

Re: SIP INFO during INVITE

To be honest, we are happy for you to say it would only be supported on the following models. We use a wide range of Polycom models from IP331 to VVX600.

 

Thank you for providing this information however this would mean the phone would have to be reached locally to send the answer curl request. Our problem is the phone could be hosted anywhere in the world but we need to send the answer message to it from our server.

 

The reason we said about SIP messages is due to having an already established connection to your phone allowing us to pass the message easily to the phone.

 

Is there anything else you could advise that would point us in the right direction?

 

Kind Regards

Warren

Message 5 of 8
SteffenBaierUK
Polycom Employee & Community Manager

Re: SIP INFO during INVITE

Hello Warren,

we only look at the Alert-Info in the SIP invite to my knowledge.

 

If this is properly formatted as highlighted in the above FAQ => here <= then we will either check against the ring class which can be multiple different options.

 

These are outlined in the above FAQ and can be autoAnswer or ringAutoAnswer or multiple other variants (Case sensitive).

 

You would need to check this against your apllication by sending an Alert-Info: info=autoAnswer and adjust the configuration example in the FAQ post during your SIP INFO.

 

If this does not work then you would need to raise this as a feature request as explained here:

 

Jan 03, 2013 Question: How can I request a change to the current Polycom SIP / UCS Software?

Resolution: Please check => here <=


Please ensure to provide some feedback if this reply has helped you so other users can profit from your experience.

Best Regards

Steffen Baier

Polycom Global Services

----------------
The title Polycom Employee & Community Manager is a community setting and does not reflect my role. I am just a simple volunteer in the community like everybody else. My official "day" Job is 3rd Level support at Poly but I am unable to provide official support via the community.

----------------

Notice: This community forum is not an official Poly support resource, thus responses from Poly employees, partners, and customers alike are best-effort in attempts to share learned knowledge. If you need immediate and/or official assistance please open a service ticket through your proper support channels.
Please also ensure you always check the VoIP , Video Endpoint , Skype for Business , PSTN or RPM FAQ's
Message 6 of 8
SteffenBaierUK
Polycom Employee & Community Manager

Re: SIP INFO during INVITE

Hello Warren,

I followed this up with one of our most senior and experienced engineers and got the following response.

 

  • In our 180 responses we list that we accept the "talk" event.  A NOTIFY on the same dialog (ie: matching to/from tags & call-id) with that event will trigger an immediate answer

Polycom 180 reply:

 

0504201613|sip  |0|00|>>> Data Send to 10.252.122.122:5060
0504201613|sip  |0|00|    SIP/2.0 180 Ringing
0504201613|sip  |0|00|    Via: SIP/2.0/UDP 10.252.122.122:5060;branch=z9hG4bK7b03f142
0504201613|sip  |0|00|    From: "Steffen 1" <sip:3080@10.252.122.122>;tag=as2ab9a3ef
0504201613|sip  |0|00|    To: "3070" <sip:3070@10.252.149.107>;tag=245EF5BF-90A06E1C
0504201613|sip  |0|00|    CSeq: 102 INVITE
0504201613|sip  |0|00|    Call-ID: 5ea2e0436ea5963a1717b93f57548480@10.252.122.122:5060
0504201613|sip  |0|00|    Contact: <sip:3070@10.252.149.107>
0504201613|sip  |0|00|    User-Agent: Polycom/5.4.1.14510 PolycomVVX-VVX_500-UA/5.4.1.14510
0504201613|sip  |0|00|    Allow-Events: conference,talk,hold
0504201613|sip  |0|00|    Accept-Language: en
0504201613|sip  |0|00|    Content-Length: 0
0504201613|sip  |0|00|    
0504201613|sip  |0|00|<<< End of data send

 

Basically sending a SIP Info

 

SIP_INFO_Event_Talk.png

 

With the Event:talk would auto answer the call.

Please ensure to provide some feedback if this reply has helped you so other users can profit from your experience.

Best Regards

Steffen Baier

Polycom Global Services

----------------
The title Polycom Employee & Community Manager is a community setting and does not reflect my role. I am just a simple volunteer in the community like everybody else. My official "day" Job is 3rd Level support at Poly but I am unable to provide official support via the community.

----------------

Notice: This community forum is not an official Poly support resource, thus responses from Poly employees, partners, and customers alike are best-effort in attempts to share learned knowledge. If you need immediate and/or official assistance please open a service ticket through your proper support channels.
Please also ensure you always check the VoIP , Video Endpoint , Skype for Business , PSTN or RPM FAQ's

View solution in original post

Message 7 of 8

Re: SIP INFO during INVITE

Thank you ever so much Steffen,

This solution worked perfectly! Once again thank you very much!

 

Regards

Warren

Message 8 of 8