[SIPForum-discussion] Sip audio negotiation problem between PBX and MCU (audioconferences)

Fri May 1 19:34:23 UTC 2015

Hi Sumedha,
I´m in contact with engineers on charge of those devices. And I´m going to explain this scenario.And I´d need to understand better how sip works.how would they change the priority of codecs configuration as preference, at sip INVITE SDP?

Does sip message by itshelf show a codec preference?In Example:when invite sdp advices codecs like this:
a=rtpmap:0 PCMU/8000
a=ptime:30
a=rtpmap:8 PCMA/8000
a=ptime:30
a=rtpmap:116 iLBC/8000
a=ptime:20

Is this message saying to remote endpoint that PCMU is the favorit, and iLBC the least in preference?
In other hands, scenario sometimes needs DSP resources to make transcoding, and it´s needed to have end points sending packets with same packetization. So we´d need endpoints to advice 30 ms.
How would they´ve to advice this packetization at SDP.I´ve seen SDP where it´s showd just the codec, but for others, it´s showed ptime or pmaxtime field.
IE:a=rtpmap:0 PCMU/8000
a=ptime:30
a=rtpmap:8 PCMA/8000
(missing ptime field) --> Does this mean PCMA adviced by endpoint could have any packetization time?, or it inherits last ptime showed for PCMU, 30ms.
a=rtpmap:116 iLBC/8000
a=ptime:20

In other hands, for pmaxtime field, if it appears just once after a codec advice, this means this advice of pmaxtime is only for codec below
In Example:
a=rtpmap:0 PCMU/8000
a=ptime:30
a=rtpmap:8 PCMA/8000
a=pmaxtime:40 ( does this field only apply to PCMA?, or remote endpoint will understand it could be used pmaxtime 40 ms only if PCMA is chosen?
a=rtpmap:116 iLBC/8000
a=ptime:20

Thank you,
regards,
Antonio      De: Sumedha Gawarikar <s.gawarikar at gmail.com>
 Para: Antonio Manchon Romero <antoniomanchonromero at yahoo.es> 
CC: discussion at sipforum.org 
 Enviado: Jueves 30 de abril de 2015 20:15
 Asunto: Re: [SIPForum-discussion] Sip audio negotiation problem between PBX and MCU (audioconferences)

It may be solved by changing the priority of codecs configuration as preference either in phones or system.Best RegardsOn Apr 30, 2015 9:41 PM, "Antonio Manchon Romero" <antoniomanchonromero at yahoo.es> wrote:

Hi,
I´d need help to troubleshoot a negotiation issue, were a phone calls to a MCU to make an audio connection.Call´s stablished at G729 30ms. And audio and dtmf tones are correct.But for some phones I´d prefer G722 or G711ulaw or G711alaw instead of G729.
I´m integrating two systems for audio calls, a sip PBX and an Proxy Server vía sip trunk. Proxy Server is a registrar server for MCU (audioconferences).

sip Phone supports: G722, PCMU, PCMA, G729 and other protocols.MCU supports: G7221, SIREN22, G729, G722, opus, PCMA, PCMU

PBX has been configured to allow any audio codec with 64kpbs as maximum bandwith compression.  And PBX has forced packetization time to 30 ms. Sip-trunk from PBX to Proxy Server is configured with early offer as needed for calls from determined places to MCU.

Call flow: phone --> PBX (sip-trunk Early Offer) --> Proxy Server --> MCU
Here I paste sip traces taken at PBX. In this example call has connected with G729 30ms, and then it´s been disconnected normally.

Some fields have been substituted by descripting names: 

Calling_Phone_Name
Calling_Numbercalled_numberPBX_signalling_ip_addressproxy_server_ip_addressMCU_signalling_ip_address
MCU_media_ip_address
INVITE sip:called_number at proxy_server_ip_address:5060 SIP/2.0
Via: SIP/2.0/UDP PBX_signalling_ip_address:5060;branch=z9hG4bK6a6ba75c4b95b
From: "Calling_Phone_Name" <sip:Calling_Number at PBX_signalling_ip_address>;tag=865212~e390a5a7-d5c3-4e13-afd1-f8f73d1be093-50102620
To: <sip:called_number at proxy_server_ip_address>
Date: Thu, 30 Apr 2015 10:10:57 GMT
Call-ID: 3032d400-5411ffb1-273ab-c180117 at PBX_signalling_ip_address
Supported: timer,resource-priority,replaces
Min-SE:  180
User-Agent: PBX
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence
Supported: X-pbx-srtp-fallback,X-pbx-original-called
Call-Info: <sip:PBX_signalling_ip_address:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
pbx-Guid: 0808637440-0000065536-0000144815-0202899735
Session-Expires:  1800
P-Asserted-Identity: "Calling_Phone_Name" <sip:Calling_Number at PBX_signalling_ip_address>
Remote-Party-ID: "Calling_Phone_Name" <sip:Calling_Number at PBX_signalling_ip_address>;party=calling;screen=yes;privacy=off
Contact: <sip:Calling_Number at PBX_signalling_ip_address:5060>
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 406

v=0
o=pbxSystemsCall_Agent-SIP 865212 1 IN IP4 PBX_signalling_ip_address
s=SIP Call
c=IN IP4 Calling_Phone_ip_address
b=TIAS:64000
b=AS:64
t=0 0
m=audio 20488 RTP/AVP 0 8 116 18 101
a=rtpmap:0 PCMU/8000
a=ptime:30
a=rtpmap:8 PCMA/8000
a=ptime:30
a=rtpmap:116 iLBC/8000
a=ptime:20
a=maxptime:100
a=fmtp:116 mode=20
a=rtpmap:18 G729/8000
a=ptime:30
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

SIP/2.0 180 Ringing
CSeq: 101 INVITE
Call-ID: 3032d400-5411ffb1-273ab-c180117 at PBX_signalling_ip_address
From: "Calling_Phone_Name" <sip:Calling_Number at PBX_signalling_ip_address>;tag=865212~e390a5a7-d5c3-4e13-afd1-f8f73d1be093-50102620
To: <sip:called_number at proxy_server_ip_address>;tag=634f26de
Via: SIP/2.0/UDP PBX_signalling_ip_address:5060;branch=z9hG4bK6a6ba75c4b95b
Content-Length: 0

SIP/2.0 200 OK
CSeq: 101 INVITE
Call-ID: 3032d400-5411ffb1-273ab-c180117 at PBX_signalling_ip_address
From: "Calling_Phone_Name" <sip:Calling_Number at PBX_signalling_ip_address>;tag=865212~e390a5a7-d5c3-4e13-afd1-f8f73d1be093-50102620
To: <sip:called_number at proxy_server_ip_address>;tag=634f26de
Via: SIP/2.0/UDP PBX_signalling_ip_address:5060;branch=z9hG4bK6a6ba75c4b95b
Allow-Events: conference,refer,conference
User-Agent: MCU_version
P-RMX-Info: m,c,128000,72,a
Allow: INVITE,ACK,BYE,CANCEL,INFO,OPTIONS,UPDATE,PRACK,SUBSCRIBE,NOTIFY,BENOTIFY
Require: timer
Supported: X-pbx-callinfo,plcm-ivr-service-provider,ms-early-media,replaces,resource-priority,histinfo,ms-safe-transfer,X-pbx-sis-5.1.0,tdialog,timer,100rel,X-pbx-srtp-fallback,ms-conf-invite,ms-sender
Contact: "inbound" <sip:called_number at proxy_server_ip_address:5060;transport=udp>
Content-Type: application/sdp
Session-Expires: 1800;refresher=uac
Content-Length: 262

v=0
o=- 1430388657 88866816 IN IP4 MCU_signalling_ip_address
s=SIP Call
c=IN IP4 MCU_media_sip_address
b=AS:8
t=0 0
m=audio 49288 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=maxptime:30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=vnd.polycom.PlcmMaskCap:0011
a=sendrecv

ACK sip:called_number at proxy_server_ip_address:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP PBX_signalling_ip_address:5060;branch=z9hG4bK6a6bd7cc5bf3c
From: "Calling_Phone_Name" <sip:Calling_Number at PBX_signalling_ip_address>;tag=865212~e390a5a7-d5c3-4e13-afd1-f8f73d1be093-50102620
To: <sip:called_number at proxy_server_ip_address>;tag=634f26de
Date: Thu, 30 Apr 2015 10:10:57 GMT
Call-ID: 3032d400-5411ffb1-273ab-c180117 at PBX_signalling_ip_address
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence
Content-Length: 0

INVITE sip:Calling_Number at PBX_signalling_ip_address:5060 SIP/2.0
Via: SIP/2.0/UDP proxy_server_ip_address:5060;branch=z9hG4bK-3739-3ca9e7aba03c8134034c251595d8e028
CSeq: 1 INVITE
From: <sip:called_number at proxy_server_ip_address>;tag=634f26de
To: "Calling_Phone_Name" <sip:Calling_Number at PBX_signalling_ip_address>;tag=865212~e390a5a7-d5c3-4e13-afd1-f8f73d1be093-50102620
Call-ID: 3032d400-5411ffb1-273ab-c180117 at PBX_signalling_ip_address
Allow-Events: conference,refer,conference
User-Agent: MCU_version
P-RMX-Info: i,c,128000,32,a
Allow: INVITE,ACK,BYE,CANCEL,INFO,OPTIONS,UPDATE,PRACK,SUBSCRIBE,NOTIFY,BENOTIFY
Supported: X-pbx-callinfo,ms-early-media,plcm-ivr-service-provider,replaces,resource-priority,histinfo,ms-safe-transfer,X-pbx-sis-5.1.0,tdialog,timer,100rel,X-pbx-srtp-fallback,ms-conf-invite,ms-sender,tdialog
Referred-By: <sip:called_number at proxy_server_ip_address:5060;/vmr;transport=UDP>
Plcm-Call-ID: 6ac0899a-0b0d-412f-852e-5e3f00db515b
Contact: "inbound" <sip:called_number at proxy_server_ip_address:5060;transport=udp>;isfocus
Max-Forwards: 69
Content-Type: application/sdp
Session-Expires: 1800;refresher=uas
Min-Expires: 90
Content-Length: 875

v=0
o=- 1430388657 88866817 IN IP4 MCU_signalling_ip_address
s=rmx2k Conf
c=IN IP4 MCU_media_sip_address
b=AS:32
t=0 0
m=audio 49290 RTP/AVP 104 105 114 113 103 102 18 9 127 8 0 98 101
a=rtpmap:104 G7221/16000
a=fmtp:104 bitrate=32000
a=rtpmap:105 SIREN22/48000
a=fmtp:105 bitrate=32000
a=rtpmap:114 G7221/32000
a=fmtp:114 bitrate=32000
a=rtpmap:113 G7221/32000
a=fmtp:113 bitrate=24000
a=rtpmap:103 G7221/16000
a=fmtp:103 bitrate=24000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=16000
a=rtpmap:18 G729/8000
a=maxptime:30
a=rtpmap:9 G722/8000
a=rtpmap:127 opus/48000/2
a=fmtp:127 sprop-maxplaybackrate=48000; maxaveragebitrate=32000; sprop-stereo=0; stereo=0
a=ptime:1
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:98 SIREN14/16000
a=fmtp:98 bitrate=32000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=vnd.polycom.PlcmMaskCap:0011
a=sendrecv

SIP/2.0 100 Trying
Via: SIP/2.0/UDP proxy_server_ip_address:5060;branch=z9hG4bK-3739-3ca9e7aba03c8134034c251595d8e028
From: <sip:called_number at proxy_server_ip_address>;tag=634f26de
To: "Calling_Phone_Name" <sip:Calling_Number at PBX_signalling_ip_address>;tag=865212~e390a5a7-d5c3-4e13-afd1-f8f73d1be093-50102620
Date: Thu, 30 Apr 2015 10:11:09 GMT
Call-ID: 3032d400-5411ffb1-273ab-c180117 at PBX_signalling_ip_address
CSeq: 1 INVITE
Allow-Events: presence
Content-Length: 0

SIP/2.0 200 OK
Via: SIP/2.0/UDP proxy_server_ip_address:5060;branch=z9hG4bK-3739-3ca9e7aba03c8134034c251595d8e028
From: <sip:called_number at proxy_server_ip_address>;tag=634f26de
To: "Calling_Phone_Name" <sip:Calling_Number at PBX_signalling_ip_address>;tag=865212~e390a5a7-d5c3-4e13-afd1-f8f73d1be093-50102620
Date: Thu, 30 Apr 2015 10:11:09 GMT
Call-ID: 3032d400-5411ffb1-273ab-c180117 at PBX_signalling_ip_address
CSeq: 1 INVITE
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
Allow-Events: presence
Supported: replaces
Supported: X-pbx-srtp-fallback
Supported: Geolocation
Session-Expires:  1800;refresher=uas
Require:  timer
P-Asserted-Identity: "Calling_Phone_Name" <sip:Calling_Number at PBX_signalling_ip_address>
Remote-Party-ID: "Calling_Phone_Name" <sip:Calling_Number at PBX_signalling_ip_address>;party=called;screen=yes;privacy=off
Contact: <sip:Calling_Number at PBX_signalling_ip_address:5060>
Content-Type: application/sdp
Content-Length: 243

v=0
o=pbxSystemsCall_Agent-SIP 865212 2 IN IP4 PBX_signalling_ip_address
s=SIP Call
c=IN IP4 Calling_Phone_ip_address
b=AS:8
t=0 0
m=audio 20488 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=ptime:30
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

ACK sip:Calling_Number at PBX_signalling_ip_address:5060 SIP/2.0
Call-ID: 3032d400-5411ffb1-273ab-c180117 at PBX_signalling_ip_address
CSeq: 1 ACK
Via: SIP/2.0/UDP proxy_server_ip_address:5060;branch=z9hG4bK-3739-a27c389a0eaf38ae5e8d6be5c2d1f79c
From: <sip:called_number at proxy_server_ip_address>;tag=634f26de
To: "Calling_Phone_Name" <sip:Calling_Number at PBX_signalling_ip_address>;tag=865212~e390a5a7-d5c3-4e13-afd1-f8f73d1be093-50102620
Max-Forwards: 70
Content-Length: 0

BYE sip:called_number at proxy_server_ip_address:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP PBX_signalling_ip_address:5060;branch=z9hG4bK6a88b1d3ee38
From: "Calling_Phone_Name" <sip:Calling_Number at PBX_signalling_ip_address>;tag=865212~e390a5a7-d5c3-4e13-afd1-f8f73d1be093-50102620
To: <sip:called_number at proxy_server_ip_address>;tag=634f26de
Date: Thu, 30 Apr 2015 10:11:09 GMT
Call-ID: 3032d400-5411ffb1-273ab-c180117 at PBX_signalling_ip_address
User-Agent: PBX
Max-Forwards: 70
P-Asserted-Identity: "Calling_Phone_Name" <sip:Calling_Number at PBX_signalling_ip_address>
CSeq: 102 BYE
Reason: Q.850;cause=16
Content-Length: 0

SIP/2.0 200 OK
CSeq: 102 BYE
Call-ID: 3032d400-5411ffb1-273ab-c180117 at PBX_signalling_ip_address
From: "Calling_Phone_Name" <sip:Calling_Number at PBX_signalling_ip_address>;tag=865212~e390a5a7-d5c3-4e13-afd1-f8f73d1be093-50102620
To: <sip:called_number at proxy_server_ip_address>;tag=634f26de
Via: SIP/2.0/UDP PBX_signalling_ip_address:5060;branch=z9hG4bK6a88b1d3ee38
Content-Length: 0
***********************************************
Thanks for your help in advance,
Regards,

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