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

Thu May 7 06:11:35 UTC 2015

Codec priority is defined by line "m="; e.g. m=... 0 8 ... means PCMU has
the highest priority followed by PCMA.

Please see RFC3264 for further details.

Antonio Manchon Romero <antoniomanchonromero at yahoo.es> 於 2015年5月2日星期六 寫道：

> 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
> <javascript:_e(%7B%7D,'cvml','s.gawarikar at gmail.com');>>
> *Para:* Antonio Manchon Romero <antoniomanchonromero at yahoo.es
> <javascript:_e(%7B%7D,'cvml','antoniomanchonromero at yahoo.es');>>
> *CC:* discussion at sipforum.org
> <javascript:_e(%7B%7D,'cvml','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 Regards
> On Apr 30, 2015 9:41 PM, "Antonio Manchon Romero" <
> antoniomanchonromero at yahoo.es
> <javascript:_e(%7B%7D,'cvml','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_Number
> called_number
> PBX_signalling_ip_address
> proxy_server_ip_address
> MCU_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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