16 Codec selecting

34.229-13GPPInternet Protocol (IP) multimedia call control protocol based on Session Initiation Protocol (SIP) and Session Description Protocol (SDP)Part 1: Protocol conformance specificationRelease 16TSUser Equipment (UE) conformance specification

Tools: ARFCN - Frequency Conversion for 5G NR/LTE/UMTS/GSM

16.1 Void

16.2 Speech AMR, indicate selective codec modes

16.2.1 Definition

Test to verify that the UE correctly performs IMS Multimedia Telephony speech call setup when selective AMR codec modes are offered. This process is described in 3GPP TS 24.173 [65], TS 24.229 [10] and TS 26.114 [66].

16.2.2 Conformance requirement

[TS 24.229, clause 5.1.4.1]

If an initial INVITE request is received the terminating UE shall check whether the terminating UE requires local resource reservation.

NOTE 1: The terminating UE can decide if local resource reservation is required based on e.g. application requirements, current access network capabilities, local configuration, etc.

If local resource reservation is required at the terminating UE and the terminating UE supports the precondition mechanism, and:

a) the received INVITE request includes the "precondition" option-tag in the Supported header or Require header, the terminating UE shall make use of the precondition mechanism and shall indicate a Require header with the "precondition" option-tag in any response or subsequent request it sends towards to the originating UE; or

…

[TS 26.114, clause 5.2.1]

MTSI terminals offering speech communication shall support:

– AMR speech codec (3GPP TS 26.071, 3GPP TS 26.090, 3GPP TS 26.073 and 3GPP TS 26.104) including all 8 modes and source controlled rate operation ‎3GPP TS 26.093. The terminal shall be capable of operating with any subset of these 8 codec modes.

[TS 24.229, clause 6.1.1]

During session establishment procedure, SIP messages shall only contain SDP payload if that is intended to modify the session description, or when the SDP payload must be included in the message because of SIP rules described in RFC 3261.

[TS 26.114, clause 6.2.5]

The SDP shall include bandwidth information for each media stream and also for the session in total. The bandwidth information for each media stream and for the session is defined by the Application Specific (AS) bandwidth modifier as defined in RFC 4566.

[TS 26.114, clause 7.3.1]

The bandwidth for RTCP traffic shall be described using the "RS" and "RR" SDP bandwidth modifiers at media level, as specified by RFC 3556.

Reference(s)

3GPP TS 24.229 [10] clause 5.1.4.1. TS 26.114 [66] clauses 5.2.1, 6.2.5, and 7.3.1.

16.2.3 Test purpose

1) To verify that, when initiating MT MTSI speech AMR call with selective codec modes and with the remote UE already having resources available, the UE performs correct exchange of SIP protocol signalling messages for setting up the session.

2) To verify that within SIP signalling the UE performs the correct exchange of SIP header and parameter contents.

3) To verify that within SIP signalling the UE performs the correct exchange of SDP contents.

4) To verify that the UE is able to release the call.

16.2.4 Method of test

Initial conditions

UE contains either ISIM and USIM applications or only USIM application on UICC. UE has activated a PDP context, discovered P-CSCF and registered to IMS services, by executing the generic test procedure in Annex C.2 or C.2a (GIBA only) up to the last step.

SS is configured with the shared secret key of IMS AKA algorithm, related to the IMS private user identity (IMPI) configured on the UICC card equipped into the UE. SS has performed AKAv1-MD5 authentication with the UE and accepted the registration (IMS security).

Test procedure

SS sends an INVITE request to the UE.
Void.

3) SS may receive 100 Trying from the UE.

3A) SS may receive 183 Session Progress from the UE.
SS triggers the activation of a dedicated bearer.

3B) SS may send PRACK to the UE to acknowledge the 183 Session Progress.

3C) SS may receive 200 OK for PRACK from the UE.

4) SS may receive 180 Ringing from the UE.

5) SS may send PRACK to the UE to acknowledge the 180 Ringing.

6) SS may receive 200 OK for PRACK from the UE.

6A) The UE accepts the session invite.
If 180 Ringing is not received from the UE after 5s from step 1, the MMI command shall be started to trigger the UE to accept the call.

7) SS expects and receives 200 OK for INVITE from the UE.

8) SS send an ACK to acknowledge receipt of the 200 OK for INVITE

9) SS sends BYE to the UE.

10) SS expects and receives 200 Ok for BYE from the UE

Expected sequence

Step	Direction		Message	Comment
	UE	SS
1	🡨		INVITE	SS sends INVITE with the first SDP offer.
2				Void
3	🡪		100 Trying	(Optional) The UE responds with a 100 Trying provisional response.
3A	🡪		183 Session Progress	(Optional) The UE sends 183 response reliably with the SDP answer to the offer in INVITE
3B	🡨		PRACK	(Optional) SS acknowledges if a 183 Session Progress is received.
3C	🡪		200 OK	(Optional) The UE responds if a PRACK is sent.
4	🡪		180 Ringing	(Optional) The UE responds to INVITE with 180 Ringing.
5	🡨		PRACK	(Optional) SS shall send PRACK if the 180 response contains 100rel option-tag in the Require header.
6	🡪		200 OK	(Optional) The UE acknowledges the PRACK with 200 OK.
6A				Make UE accept the speech AMR offer.
7	🡪		200 OK	The UE responds INVITE with 200 OK.
8	🡨		ACK	The SS acknowledges the receipt of 200 OK for INVITE.
9	🡨		BYE	The SS releases the call with BYE.
10	🡪		200 OK	The UE sends 200 OK for BYE.

NOTE 1: The default messages contents in annex A are used with condition “IMS security” or “GIBA” when applicable

NOTE 2: Steps 4, 5, and 6 can happen in parallel to steps 3B and 3C

Specific Message Contents

INVITE (Step 1)

Use the default message "INVITE for MT Call" in annex A.2.9, with the following exceptions:

Header/param	Value/Remark
Supported
option-tag	precondition
Message-body	The following SDP types and values. Session description: v=0 o=- 1111111111 1111111111 IN (addrtype) (unicast-address for SS) s=- c=IN (addrtype) (connection-address for SS) b=AS:37 Time description: t=0 0 Media description: m=audio (transport port) RTP/AVP 99 100 b=AS:37 b=RS:0 b=RR:2000 Attributes for media: a=rtpmap:99 AMR/8000/1 a=fmtp:99 mode-set=0,2,4,7; mode-change-capability=2; max-red=220 a=rtpmap: 100 telephone-event/8000 a=fmtp: 100 0-15 a=ptime:20 a=maxptime:240 Attributes for preconditions: a=curr:qos local sendrecv a=curr:qos remote none a=des:qos mandatory local sendrecv a=des:qos optional remote sendrecv

100 Trying for INVITE (Step 3)

183 Session Progress (Step 3A)

Use the default message "183 Session Progress" in annex A.2.3 with the following exceptions:

Header/param	Value/remark
Status-Line
Reason-Phrase	Not checked
Require
option-tag	precondition
Message-body	The following SDP types and values shall be present. Session description: v=0 o=(username) (sess-id) (sess-version) IN (addrtype) (unicast-address for UE) s=(session name) c=IN (addrtype) (connection-address for UE) [Note 1] b=AS: (bandwidth-value) Time description: t=0 0 Media description: m=audio (transport port) RTP/AVP (fmt) c=IN (addrtype) (connection-address for UE) [Note 1] b=AS: (bandwidth-value) b=RS: (bandwidth-value) b=RR: (bandwidth-value) Attributes for media: a=rtpmap:(payload type) AMR/8000 [Note 2] a=fmtp:(format) Attributes for preconditions: a=curr:qos local none a=curr:qos remote sendrecv a=des:qos mandatory local sendrecv a=des:qos mandatory remote sendrecv Note 1: At least one "c=" field shall be present. Note 2: The AMR channel number shall be “/1” or omitted.

180 Ringing (Step 4)

Use the default message “180 Ringing for INVITE” in annex A.2.6 with the following exceptions:

Header/param	Value/remark
Content-Type	Header optional Contents if present:
media-type	application/sdp
Content-Length	header shall be present if UE uses TCP to send this message and if there is a message body
value	length of message-body
Message-body	optional if 183 Session Progress is not used not present if 183 Session Progress is used (step 3A) Contents if present: The following SDP types and values shall be present. Session description: v=0 o=(username) (sess-id) (sess-version) IN (addrtype) (unicast-address for UE) s=(session name) c=IN (addrtype) (connection-address for UE) [Note 1] b=AS: (bandwidth-value) Time description: t=0 0 Media description: m=audio (transport port) RTP/AVP (fmt) c=IN (addrtype) (connection-address for UE) [Note 1] b=AS: (bandwidth-value) b=RS: (bandwidth-value) b=RR: (bandwidth-value) Attributes for media: a=rtpmap:(payload type) AMR/8000 [Note 2] a=fmtp:(format) mode-set=0,2,4,7; Attributes for preconditions: a=curr:qos local sendrecv a=curr:qos remote sendrecv a=des:qos mandatory local sendrecv a=des:qos mandatory remote sendrecv Note 1: At least one "c=" field shall be present. Note 2: The AMR channel number shall be “/1” or omitted.

200 OK for INVITE (Step 7)

Use the default message “200 OK for other requests than REGISTER or SUBSCRIBE” in annex A.3.1 with the following exceptions:

Header/param	Value/remark
Content-Type	Header optional Contents if present:
media-type	application/sdp
Content-Length	header shall be present if UE uses TCP to send this message and if there is a message body
value	length of message-body
Message-body	not present if 183 Session Progress is used (step 3A) or 180 Ringing (step 4) contained SDP. present if 183 Session Progress is not used (step 3A) and 180 Ringing (step 4) did not contain SDP. Contents if present: The same requirements for SDP types and values as specified in step 4.

16.2.5 Test requirements

The UE shall send requests and responses as described in clause 16.2.4.

16.3 Speech AMR-WB, indicate all codec modes

16.3.1 Definition

Test to verify that the UE correctly performs IMS Multimedia Telephony speech call setup when all AMR-WB codec modes are offered. This process is described in 3GPP TS 24.173 [65], TS 24.229 [10] and TS 26.114 [66].

16.3.2 Conformance requirement

[TS 24.229, clause 5.1.4.1]

If an initial INVITE request is received the terminating UE shall check whether the terminating UE requires local resource reservation.

NOTE 1: The terminating UE can decide if local resource reservation is required based on e.g. application requirements, current access network capabilities, local configuration, etc.

If local resource reservation is required at the terminating UE and the terminating UE supports the precondition mechanism, and:

…

[TS 26.114, clause 5.2.1]

MTSI terminals offering speech communication shall support:

…

MTSI terminals offering wideband speech communication at 16 kHz sampling frequency shall support:

– AMR wideband codec (3GPP TS 26.171, 3GPP TS 26.190, 3GPP TS 26.173 and 3GPP TS 26.204) including all 9 modes and source controlled rate operation ‎3GPP TS 26.193. The terminal shall be capable of operating with any subset of these 9 codec modes.

…

MTSI terminals offering wideband speech communication shall also offer narrowband speech communications. When offering both wideband speech and narrowband speech communication, wideband shall be listed as the first payload type in the m line of the SDP offer (RFC 4566).

[TS 24.229, clause 6.1.1]

[TS 26.114, clause 6.2.5]

[TS 26.114, clause 7.3.1]

The bandwidth for RTCP traffic shall be described using the "RS" and "RR" SDP bandwidth modifiers at media level, as specified by RFC 3556.

Reference(s)

3GPP TS 24.229 [10] clause 5.1.4.1, TS 26.114 [66] clauses 5.2.1, 6.2.5, and 7.3.1.

16.3.3 Test purpose

1) To verify that, when initiating MT MTSI speech AMR-WB call and with the remote UE already having resources available, the UE performs correct exchange of SIP protocol signalling messages for setting up the session.

2) To verify that within SIP signalling the UE performs the correct exchange of SIP header and parameter contents.

3) To verify that within SIP signalling the UE performs the correct exchange of SDP contents.

4) To verify that the UE is able to release the call.

16.3.4 Method of test

Initial conditions

Test procedure

1) SS sends an INVITE request to the UE.

2) Void.

3) SS may receive 100 Trying from the UE.

4) SS may receive 183 Session Progress from the UE.
SS triggers the activation of a dedicated bearer.

5) SS may send PRACK to the UE to acknowledge the 183 Session Progress.

6) SS may receive 200 OK for PRACK from the UE.

7) Void.

8) Void.

9) SS may receive 180 Ringing from the UE.

10) SS may send PRACK to the UE to acknowledge the 180 Ringing.

11) SS may receive 200 OK for PRACK from the UE.

11A) The UE accepts the session invite.
If 180 Ringing is not received from the UE after 5s from step 1, the MMI command shall be started to trigger the UE to accept the call.

12) SS expects and receives 200 OK for INVITE from the UE.

13) SS send an ACK to acknowledge receipt of the 200 OK for INVITE

14) SS sends BYE to the UE.

15) SS expects and receives 200 Ok for BYE from the UE

Expected sequence

Step	Direction		Message	Comment
	UE	SS
1	🡨		INVITE	SS sends INVITE with the first SDP offer.
2				Void
3	🡪		100 Trying	(Optional) The UE responds with a 100 Trying provisional response.
4	🡪		183 Session Progress	(Optional) The UE sends 183 response reliably with the SDP answer to the offer in INVITE
5	🡨		PRACK	(Optional) SS acknowledges if a 183 Session Progress is received.
6	🡪		200 OK	(Optional) The UE responds if a PRACK is sent.
7	🡨		Void
8	🡪		Void
9	🡪		180 Ringing	(Optional) The UE responds to INVITE with 180 Ringing.
10	🡨		PRACK	(Optional) SS shall send PRACK if the 180 response contains 100rel option-tag in the Require header.
11	🡪		200 OK	(Optional) The UE acknowledges the PRACK with 200 OK.
11A				Make UE accept the speech AMR WB offer.
12	🡪		200 OK	The UE responds INVITE with 200 OK.
13	🡨		ACK	The SS acknowledges the receipt of 200 OK for INVITE.
14	🡨		BYE	The SS releases the call with BYE.
15	🡪		200 OK	The UE sends 200 OK for BYE.

NOTE 1: The default messages contents in annex A are used with condition “IMS security“ or “GIBA” when applicable.

NOTE 2: Steps 9, 10, and 11 can happen in parallel to steps 5 and 6.

Specific Message Contents

INVITE (Step 1)

Use the default message "INVITE for MT Call" in annex A.2.9, with the following exceptions:

Header/param	Value/Remark
Supported
option-tag	precondition
Message-body	The following SDP types and values. Session description: v=0 o=- 1111111111 1111111111 IN (addrtype) (unicast-address for SS) s=- c=IN (addrtype) (connection-address for SS) b=AS:49 Time description: t=0 0 Media description: m=audio (transport port) RTP/AVP 97 98 99 100 b=AS:49 b=RS:0 b=RR:2000 Attributes for media: a=rtpmap:97 AMR-WB/16000/1 a=fmtp:97 mode-change-capability=2; max-red=220 a=rtpmap: 98 telephone-event/16000 a=fmtp: 98 0-15 a=rtpmap:99 AMR/8000/1 a=fmtp:99 mode-change-capability=2; max-red=220 a=rtpmap: 100 telephone-event/8000 a=fmtp: 100 0-15 a=ptime:20 a=maxptime:240 Attributes for preconditions: a=curr:qos local sendrecv a=curr:qos remote none a=des:qos mandatory local sendrecv a=des:qos optional remote sendrecv

100 Trying for INVITE (Step 3)

Use the default message “100 Trying for INVITE” in annex A.2.2

183 Session Progress (Step 4)

Use the default message "183 Session Progress" in annex A.2.3 with the following exceptions:

Header/param	Value/remark
Status-Line
Reason-Phrase	Not checked
Require
option-tag	precondition
Message-body	The following SDP types and values shall be present. Session description: v=0 o=(username) (sess-id) (sess-version) IN (addrtype) (unicast-address for UE) s=(session name) c=IN (addrtype) (connection-address for UE) [Note 1] b=AS: (bandwidth-value) Time description: t=0 0 Media description: m=audio (transport port) RTP/AVP (fmt) c=IN (addrtype) (connection-address for UE) [Note 1] b=AS: (bandwidth-value) b=RS: (bandwidth-value) b=RR: (bandwidth-value) Attributes for media: a=rtpmap:(payload type) AMR-WB/16000 [Note 2] a=fmtp:(format) Attributes for preconditions: a=curr: a=curr:qos local none a=curr:qos remote sendrecv a=des:qos mandatory local sendrecv a=des:qos mandatory remote sendrecv Note 1: At least one "c=" field shall be present. Note 2: The AMR channel number shall be “/1” or omitted.

PRACK (step 5)

Use the default message "PRACK" in annex A.2.4. No content body is included in this PRACK message.

200 OK (Step 6)

Use the default message "200 OK for other requests than REGISTER or SUBSCRIBE" in annex A.3.1.

180 Ringing (Step 9)

Use the default message “180 Ringing for INVITE” in annex A.2.6 with the following exceptions:

Header/param	Value/remark
Content-Type	Header optional Contents if present:
media-type	application/sdp
Content-Length	header shall be present if UE uses TCP to send this message and if there is a message body
value	length of message-body
Message-body	optional if 183 Session Progress is not used not present if 183 Session Progress is used (step 4) Contents if present: The following SDP types and values shall be present. Session description: v=0 o=(username) (sess-id) (sess-version) IN (addrtype) (unicast-address for UE) s=(session name) c=IN (addrtype) (connection-address for UE) [Note 1] b=AS: (bandwidth-value) Time description: t=0 0 Media description: m=audio (transport port) RTP/AVP (fmt) c=IN (addrtype) (connection-address for UE) [Note 1] b=AS: (bandwidth-value) b=RS: (bandwidth-value) b=RR: (bandwidth-value) Attributes for media: a=rtpmap:(payload type) AMR-WB/16000 [Note 2] a=fmtp:(format) Attributes for preconditions: a=curr:qos local sendrecv a=curr:qos remote sendrecv a=des:qos mandatory local sendrecv a=des:qos mandatory remote sendrecv Note 1: At least one "c=" field shall be present. Note 2: The AMR channel number shall be “/1” or omitted.

PRACK (step 10)

Use the default message "PRACK" in annex A.2.4. No content body is included in this PRACK message

200 OK (Step 11)

Use the default message “200 OK for other requests than REGISTER or SUBSCRIBE” in annex A.3.1.

200 OK for INVITE (Step 12)

Use the default message “200 OK for other requests than REGISTER or SUBSCRIBE” in annex A.3.1 with the following exceptions:

Header/param	Value/remark
Content-Type	Header optional Contents if present:
media-type	application/sdp
Content-Length	header shall be present if UE uses TCP to send this message and if there is a message body
value	length of message-body
Message-body	not present if 183 Session Progress is used (step 4) or 180 Ringing (step 9) contained SDP. present if 183 Session Progress is not used (step 4) and 180 Ringing (step 9) did not contain SDP. Contents if present: The same requirements for SDP types and values as specified in step 9.

ACK (Step 13)

Use the default message “ACK” in annex A.2.7.

BYE (step 14)

Use the default message "BYE" in annex A.2.8.

200 OK (step 15)

Use the default message "200 OK for other requests than REGISTER or SUBSCRIBE" in annex A.3.1.

16.3.5 Test requirements

The UE shall send requests and responses as described in clause 16.3.4.

16.4 Speech AMR-WB, indicate selective codec modes

16.4.1 Definition

Test to verify that the UE correctly performs IMS Multimedia Telephony speech call setup when selective AMR-WB codec modes are offered. This process is described in 3GPP TS 24.173 [65], TS 24.229 [10] and TS 26.114 [66].

16.4.2 Conformance requirement

Same as 34.229-1 clause 16.3.2.

16.4.3 Test purpose

1) To verify that, when initiating MT MTSI speech AMR-WB call with selective codec modes and with the remote UE already having resources available, the UE performs correct exchange of SIP protocol signalling messages for setting up the session.

2) To verify that within SIP signalling the UE performs the correct exchange of SIP header and parameter contents.

3) To verify that within SIP signalling the UE performs the correct exchange of SDP contents.

4) To verify that the UE is able to release the call.

16.4.4 Method of test

Same as 34.229-1 clause 16.3.4 except

Specific Message Contents

INVITE (Step 1)

Use the default message "INVITE for MT Call" in annex A.2.9, with the following exceptions:

Header/param	Value/Remark
Supported
option-tag	precondition
Message-body	The following SDP types and values. Session description: v=0 o=- 1111111111 1111111111 IN (addrtype) (unicast-address for SS) s=- b=AS:38 Time description: t=0 0 Media description: m=audio (transport port) RTP/AVP 97 98 99 100 c= IN (addrtype) (connection-address for SS) b=AS:38 b=RS:0 b=RR:2000 Attributes for media: a=rtpmap:97 AMR-WB/16000/1 a=fmtp:97 mode-set=0,1,2; mode-change-capability=2; max-red=220 a=rtpmap: 98 telephone-event/16000 a=fmtp: 98 0-15 a=rtpmap:99 AMR/8000/1 a=fmtp:99 mode-set=0,2,4,7; mode-change-capability=2; max-red=220 a=rtpmap: 100 telephone-event/8000 a=fmtp: 100 0-15 a=ptime:20 a=maxptime:240 Attributes for preconditions: a=curr:qos local sendrecv a=curr:qos remote none a=des:qos mandatory local sendrecv a=des:qos optional remote sendrecv

183 Session Progress (Step 4)

Use the default message "183 Session Progress" in annex A.2.3 with the following exceptions:

Header/param	Value/remark
Status-Line
Reason-Phrase	Not checked
Require
option-tag	precondition
Message-body	The following SDP types and values shall be present. Session description: v=0 o=(username) (sess-id) (sess-version) IN (addrtype) (unicast-address for UE) s=(session name) c=IN (addrtype) (connection-address for UE) [Note 1] b=AS: (bandwidth-value) Time description: t=0 0 Media description: m=audio (transport port) RTP/AVP (fmt) c=IN (addrtype) (connection-address for UE) [Note 1] b=AS: (bandwidth-value) b=RS: (bandwidth-value) b=RR: (bandwidth-value) Attributes for media: a=rtpmap: (payload type) AMR-WB/16000 [Note 2] a=fmtp: (format) mode-set=0,1,2; Attributes for preconditions: a=a=curr:qos local none a=curr:qos remote sendrecv a=des:qos mandatory local sendrecv a=des:qos mandatory remote sendrecv Note 1: At least one "c=" field shall be present. Note 2: The AMR channel number shall be “/1” or omitted.

180 Ringing (Step 9)

Use the default message “180 Ringing for INVITE” in annex A.2.6 with the following exceptions:

Header/param	Value/remark
Content-Type	Header optional Contents if present:
media-type	application/sdp
Content-Length	header shall be present if UE uses TCP to send this message and if there is a message body
value	length of message-body
Message-body	optional if 183 Session Progress is not used not present if 183 Session Progress is used (step 4) Contents if present: The following SDP types and values shall be present. Session description: v=0 o=(username) (sess-id) (sess-version) IN (addrtype) (unicast-address for UE) s=(session name) c=IN (addrtype) (connection-address for UE) [Note 1] b=AS: (bandwidth-value) Time description: t=0 0 Media description: m=audio (transport port) RTP/AVP (fmt) c= IN (addrtype) (connection-address for UE) [Note 1] b=AS: (bandwidth-value) b=RS: (bandwidth-value) b=RR: (bandwidth-value) Attributes for media: a=rtpmap:(payload type) AMR-WB/16000 [Note 2] a=fmtp:(format) mode-set=0,1,2; Attributes for preconditions: a=curr:qos local sendrecv a=curr:qos remote sendrecv a=des:qos mandatory local sendrecv a=des:qos mandatory remote sendrecv Note 1: At least one "c=" field shall be present. Note 2: The AMR channel number shall be “/1” or omitted.

200 OK for INVITE (Step 12)

Use the default message “200 OK for other requests than REGISTER or SUBSCRIBE” in annex A.3.1 with the following exceptions:

Header/param	Value/remark
Content-Type	Header optional Contents if present:
media-type	application/sdp
Content-Length	header shall be present if UE uses TCP to send this message and if there is a message body
value	length of message-body
Message-body	not present if 183 Session Progress is used (step 4) or 180 Ringing (step 9) contained SDP. present if 183 Session Progress is not used (step 4) and 180 Ringing (step 9) did not contain SDP. Contents if present: The same requirements for SDP types and values as specified in step 9.