[asterisk-users] Cisco AS5300 and Digium g729A codec
Hi, We have a problem connecting to a Cisco AS5300 trunk. We set the sip peer to allow only g729. The call attempt is able to connect, but when answered, no audio is heard or transmitted. Our asterisk version is 1.6.2.14 . Codec is licensed, bought from Digium. We do not have this problem on our other providers using asterisk and other non-cisco systems. Anyone else having this same problem? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco AS5300 and Digium g729A codec
You are hereby encouraged to post your AS5300 IOS config, sip.conf peer declaration, and packet capture. Those three things would aid greatly in diagnosis, especially the capture. -- This message was painstakingly thumbed out on my mobile, so apologies for brevity, errors, and general sloppiness. Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ On Jan 9, 2012, at 3:20 AM, Roi Stork roi.st...@gmail.com wrote: Hi, We have a problem connecting to a Cisco AS5300 trunk. We set the sip peer to allow only g729. The call attempt is able to connect, but when answered, no audio is heard or transmitted. Our asterisk version is 1.6.2.14 . Codec is licensed, bought from Digium. We do not have this problem on our other providers using asterisk and other non-cisco systems. Anyone else having this same problem? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco AS5300 and Digium g729A codec
Hi Alex, here's the config and the sip debug output. Guide: xxx.xxx.xxx.xxx - the provider's cisco as5300 trunk ip add yyy.yy.yy.yy - our asterisk 1.6.2.14 server sip config: type=peer disallow=all allow=g729 host=xxx.xxx.xxx.xxx fromdomain=xxx.xxx.xxx.xxx dtmfmode=rfc2833 nat=no canreinvite=yes context=from-trunk-sip-iaccess sip debug: v=0 o=root 249777024 249777024 IN IP4 yyy.yy.yy.yy s=Asterisk PBX 1.6.2.14 c=IN IP4 yyy.yy.yy.yy t=0 0 m=audio 13702 RTP/AVP 0 8 3 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- --- SIP read from UDP:xxx.xxx.xxx.xxx:5060 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP yyy.yy.yy.yy:5060;branch=z9hG4bK60c02567;rport From: 6598715968 sip:6598715...@yyy.yy.yy.yy;tag=as6e218907 To: sip:34546598715...@xxx.xxx.xxx.xxx Date: Fri, 06 Jan 2012 04:51:39 GMT Call-ID: 7e54da423b0e6e457475ab17694e5...@yyy.yy.yy.yy Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow-Events: telephone-event Content-Length: 0 - --- (10 headers 0 lines) --- Retransmitting #3 (no NAT) to zzz.zz.zz.zz:5060: OPTIONS sip:zzz.zz.zz.zz SIP/2.0 Via: SIP/2.0/UDP yyy.yy.yy.yy:5060;branch=z9hG4bK78129860;rport Max-Forwards: 70 From: Unknown sip:unkn...@yyy.yy.yy.yy;tag=as5c8e3f97 To: sip:zzz.zz.zz.zz Contact: sip:unkn...@yyy.yy.yy.yy Call-ID: 7bdb028f789e3afa58b22db472d9d...@yyy.yy.yy.yy CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.14 Date: Fri, 06 Jan 2012 06:23:00 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- --- SIP read from UDP:69.90.209.57:5060 --- - Retransmitting #4 (no NAT) to zzz.zz.zz.zz:5060: OPTIONS sip:zzz.zz.zz.zz SIP/2.0 Via: SIP/2.0/UDP yyy.yy.yy.yy:5060;branch=z9hG4bK78129860;rport Max-Forwards: 70 From: Unknown sip:unkn...@yyy.yy.yy.yy;tag=as5c8e3f97 To: sip:zzz.zz.zz.zz Contact: sip:unkn...@yyy.yy.yy.yy Call-ID: 7bdb028f789e3afa58b22db472d9d...@yyy.yy.yy.yy CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.14 Date: Fri, 06 Jan 2012 06:23:00 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Really destroying SIP dialog '7bdb028f789e3afa58b22db472d9d...@yyy.yy.yy.yy' Method: OPTIONS --- SIP read from UDP:xxx.xxx.xxx.xxx:5060 --- SIP/2.0 183 Session Progress Via: SIP/2.0/UDP yyy.yy.yy.yy:5060;branch=z9hG4bK60c02567;rport From: 6598715968 sip:6598715...@yyy.yy.yy.yy;tag=as6e218907 To: sip:34546598715...@xxx.xxx.xxx.xxx;tag=B6534850-EC6 Date: Fri, 06 Jan 2012 04:51:39 GMT Call-ID: 7e54da423b0e6e457475ab17694e5...@yyy.yy.yy.yy Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events: telephone-event Remote-Party-ID: 6598715968 sip:1234#6598715...@xxx.xxx.xxx.xxx;party=called;screen=no;privacy=off Contact: sip:34546598715...@xxx.xxx.xxx.xxx:5060 Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 223 v=0 o=CiscoSystemsSIP-GW-UserAgent 6911 3862 IN IP4 xxx.xxx.xxx.xxx s=SIP Call c=IN IP4 xxx.xxx.xxx.xxx t=0 0 m=audio 18132 RTP/AVP 18 c=IN IP4 xxx.xxx.xxx.xxx a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=ptime:20 - --- (15 headers 10 lines) --- Found RTP audio format 18 Found audio description format G729 for ID 18 Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port xxx.xxx.xxx.xxx:18132 --- SIP read from UDP:xxx.xxx.xxx.xxx:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP yyy.yy.yy.yy:5060;branch=z9hG4bK60c02567;rport From: 6598715968 sip:6598715...@yyy.yy.yy.yy;tag=as6e218907 To: sip:34546598715...@xxx.xxx.xxx.xxx;tag=B6534850-EC6 Date: Fri, 06 Jan 2012 04:51:39 GMT Call-ID: 7e54da423b0e6e457475ab17694e5...@yyy.yy.yy.yy Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events: telephone-event Contact: sip:34546598715...@xxx.xxx.xxx.xxx:5060 Supported: replaces Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 223 v=0 o=CiscoSystemsSIP-GW-UserAgent 6911 3862 IN IP4 xxx.xxx.xxx.xxx s=SIP Call c=IN IP4 xxx.xxx.xxx.xxx t=0 0 m=audio 18132 RTP/AVP 18 c=IN IP4 xxx.xxx.xxx.xxx a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=ptime:20 - --- (15 headers 10 lines) --- list_route: hop: sip:34546598715...@xxx.xxx.xxx.xxx:5060 set_destination: Parsing sip:34546598715...@xxx.xxx.xxx.xxx:5060 for address/port to send to set_destination: set destination to xxx.xxx.xxx.xxx, port 5060 Transmitting (no NAT) to xxx.xxx.xxx.xxx:5060: ACK sip:34546598715...@xxx.xxx.xxx.xxx:5060
Re: [asterisk-users] Cisco AS5300 and Digium g729A codec
Here's the cisco AS5300 settings from our provider codec preference 1 g729r8 codec preference 2 g729br8 codec preference 3 g723r53 codec preference 4 g723r63 codec preference 5 g723ar53 codec preference 6 g723ar63 On Mon, Jan 9, 2012 at 5:18 PM, Roi Stork roi.st...@gmail.com wrote: Hi Alex, here's the config and the sip debug output. Guide: xxx.xxx.xxx.xxx - the provider's cisco as5300 trunk ip add yyy.yy.yy.yy - our asterisk 1.6.2.14 server sip config: type=peer disallow=all allow=g729 host=xxx.xxx.xxx.xxx fromdomain=xxx.xxx.xxx.xxx dtmfmode=rfc2833 nat=no canreinvite=yes context=from-trunk-sip-iaccess sip debug: v=0 o=root 249777024 249777024 IN IP4 yyy.yy.yy.yy s=Asterisk PBX 1.6.2.14 c=IN IP4 yyy.yy.yy.yy t=0 0 m=audio 13702 RTP/AVP 0 8 3 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- --- SIP read from UDP:xxx.xxx.xxx.xxx:5060 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP yyy.yy.yy.yy:5060;branch=z9hG4bK60c02567;rport From: 6598715968 sip:6598715...@yyy.yy.yy.yy;tag=as6e218907 To: sip:34546598715...@xxx.xxx.xxx.xxx Date: Fri, 06 Jan 2012 04:51:39 GMT Call-ID: 7e54da423b0e6e457475ab17694e5...@yyy.yy.yy.yy Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow-Events: telephone-event Content-Length: 0 - --- (10 headers 0 lines) --- Retransmitting #3 (no NAT) to zzz.zz.zz.zz:5060: OPTIONS sip:zzz.zz.zz.zz SIP/2.0 Via: SIP/2.0/UDP yyy.yy.yy.yy:5060;branch=z9hG4bK78129860;rport Max-Forwards: 70 From: Unknown sip:unkn...@yyy.yy.yy.yy;tag=as5c8e3f97 To: sip:zzz.zz.zz.zz Contact: sip:unkn...@yyy.yy.yy.yy Call-ID: 7bdb028f789e3afa58b22db472d9d...@yyy.yy.yy.yy CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.14 Date: Fri, 06 Jan 2012 06:23:00 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- --- SIP read from UDP:69.90.209.57:5060 --- - Retransmitting #4 (no NAT) to zzz.zz.zz.zz:5060: OPTIONS sip:zzz.zz.zz.zz SIP/2.0 Via: SIP/2.0/UDP yyy.yy.yy.yy:5060;branch=z9hG4bK78129860;rport Max-Forwards: 70 From: Unknown sip:unkn...@yyy.yy.yy.yy;tag=as5c8e3f97 To: sip:zzz.zz.zz.zz Contact: sip:unkn...@yyy.yy.yy.yy Call-ID: 7bdb028f789e3afa58b22db472d9d...@yyy.yy.yy.yy CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.14 Date: Fri, 06 Jan 2012 06:23:00 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Really destroying SIP dialog '7bdb028f789e3afa58b22db472d9d...@yyy.yy.yy.yy' Method: OPTIONS --- SIP read from UDP:xxx.xxx.xxx.xxx:5060 --- SIP/2.0 183 Session Progress Via: SIP/2.0/UDP yyy.yy.yy.yy:5060;branch=z9hG4bK60c02567;rport From: 6598715968 sip:6598715...@yyy.yy.yy.yy;tag=as6e218907 To: sip:34546598715...@xxx.xxx.xxx.xxx;tag=B6534850-EC6 Date: Fri, 06 Jan 2012 04:51:39 GMT Call-ID: 7e54da423b0e6e457475ab17694e5...@yyy.yy.yy.yy Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events: telephone-event Remote-Party-ID: 6598715968 sip:1234#6598715...@xxx.xxx.xxx.xxx;party=called;screen=no;privacy=off Contact: sip:34546598715...@xxx.xxx.xxx.xxx:5060 Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 223 v=0 o=CiscoSystemsSIP-GW-UserAgent 6911 3862 IN IP4 xxx.xxx.xxx.xxx s=SIP Call c=IN IP4 xxx.xxx.xxx.xxx t=0 0 m=audio 18132 RTP/AVP 18 c=IN IP4 xxx.xxx.xxx.xxx a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=ptime:20 - --- (15 headers 10 lines) --- Found RTP audio format 18 Found audio description format G729 for ID 18 Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port xxx.xxx.xxx.xxx:18132 --- SIP read from UDP:xxx.xxx.xxx.xxx:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP yyy.yy.yy.yy:5060;branch=z9hG4bK60c02567;rport From: 6598715968 sip:6598715...@yyy.yy.yy.yy;tag=as6e218907 To: sip:34546598715...@xxx.xxx.xxx.xxx;tag=B6534850-EC6 Date: Fri, 06 Jan 2012 04:51:39 GMT Call-ID: 7e54da423b0e6e457475ab17694e5...@yyy.yy.yy.yy Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events: telephone-event Contact: sip:34546598715...@xxx.xxx.xxx.xxx:5060 Supported: replaces Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 223 v=0 o=CiscoSystemsSIP-GW-UserAgent 6911 3862 IN IP4 xxx.xxx.xxx.xxx s=SIP Call c=IN IP4 xxx.xxx.xxx.xxx t=0 0 m=audio 18132 RTP/AVP 18
Re: [asterisk-users] Cisco AS5300 and Digium g729A codec
The problem has been fixed. We are able to hear audio in our calls after adding these lines in the AS5300 config: sip-ua g729-annexb override There's an issue regarding codec matching in IOS versions 12.3(18) or higher: https://supportforums.cisco.com/docs/DOC-3186 On Tue, Jan 10, 2012 at 10:30 AM, Roi Stork roi.st...@gmail.com wrote: Here's the cisco AS5300 settings from our provider codec preference 1 g729r8 codec preference 2 g729br8 codec preference 3 g723r53 codec preference 4 g723r63 codec preference 5 g723ar53 codec preference 6 g723ar63 On Mon, Jan 9, 2012 at 5:18 PM, Roi Stork roi.st...@gmail.com wrote: Hi Alex, here's the config and the sip debug output. Guide: xxx.xxx.xxx.xxx - the provider's cisco as5300 trunk ip add yyy.yy.yy.yy - our asterisk 1.6.2.14 server sip config: type=peer disallow=all allow=g729 host=xxx.xxx.xxx.xxx fromdomain=xxx.xxx.xxx.xxx dtmfmode=rfc2833 nat=no canreinvite=yes context=from-trunk-sip-iaccess sip debug: v=0 o=root 249777024 249777024 IN IP4 yyy.yy.yy.yy s=Asterisk PBX 1.6.2.14 c=IN IP4 yyy.yy.yy.yy t=0 0 m=audio 13702 RTP/AVP 0 8 3 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- --- SIP read from UDP:xxx.xxx.xxx.xxx:5060 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP yyy.yy.yy.yy:5060;branch=z9hG4bK60c02567;rport From: 6598715968 sip:6598715...@yyy.yy.yy.yy;tag=as6e218907 To: sip:34546598715...@xxx.xxx.xxx.xxx Date: Fri, 06 Jan 2012 04:51:39 GMT Call-ID: 7e54da423b0e6e457475ab17694e5...@yyy.yy.yy.yy Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow-Events: telephone-event Content-Length: 0 - --- (10 headers 0 lines) --- Retransmitting #3 (no NAT) to zzz.zz.zz.zz:5060: OPTIONS sip:zzz.zz.zz.zz SIP/2.0 Via: SIP/2.0/UDP yyy.yy.yy.yy:5060;branch=z9hG4bK78129860;rport Max-Forwards: 70 From: Unknown sip:unkn...@yyy.yy.yy.yy;tag=as5c8e3f97 To: sip:zzz.zz.zz.zz Contact: sip:unkn...@yyy.yy.yy.yy Call-ID: 7bdb028f789e3afa58b22db472d9d...@yyy.yy.yy.yy CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.14 Date: Fri, 06 Jan 2012 06:23:00 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- --- SIP read from UDP:69.90.209.57:5060 --- - Retransmitting #4 (no NAT) to zzz.zz.zz.zz:5060: OPTIONS sip:zzz.zz.zz.zz SIP/2.0 Via: SIP/2.0/UDP yyy.yy.yy.yy:5060;branch=z9hG4bK78129860;rport Max-Forwards: 70 From: Unknown sip:unkn...@yyy.yy.yy.yy;tag=as5c8e3f97 To: sip:zzz.zz.zz.zz Contact: sip:unkn...@yyy.yy.yy.yy Call-ID: 7bdb028f789e3afa58b22db472d9d...@yyy.yy.yy.yy CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.14 Date: Fri, 06 Jan 2012 06:23:00 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Really destroying SIP dialog '7bdb028f789e3afa58b22db472d9d...@yyy.yy.yy.yy' Method: OPTIONS --- SIP read from UDP:xxx.xxx.xxx.xxx:5060 --- SIP/2.0 183 Session Progress Via: SIP/2.0/UDP yyy.yy.yy.yy:5060;branch=z9hG4bK60c02567;rport From: 6598715968 sip:6598715...@yyy.yy.yy.yy;tag=as6e218907 To: sip:34546598715...@xxx.xxx.xxx.xxx;tag=B6534850-EC6 Date: Fri, 06 Jan 2012 04:51:39 GMT Call-ID: 7e54da423b0e6e457475ab17694e5...@yyy.yy.yy.yy Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events: telephone-event Remote-Party-ID: 6598715968 sip:1234#6598715...@xxx.xxx.xxx.xxx;party=called;screen=no;privacy=off Contact: sip:34546598715...@xxx.xxx.xxx.xxx:5060 Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 223 v=0 o=CiscoSystemsSIP-GW-UserAgent 6911 3862 IN IP4 xxx.xxx.xxx.xxx s=SIP Call c=IN IP4 xxx.xxx.xxx.xxx t=0 0 m=audio 18132 RTP/AVP 18 c=IN IP4 xxx.xxx.xxx.xxx a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=ptime:20 - --- (15 headers 10 lines) --- Found RTP audio format 18 Found audio description format G729 for ID 18 Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port xxx.xxx.xxx.xxx:18132 --- SIP read from UDP:xxx.xxx.xxx.xxx:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP yyy.yy.yy.yy:5060;branch=z9hG4bK60c02567;rport From: 6598715968 sip:6598715...@yyy.yy.yy.yy;tag=as6e218907 To: sip:34546598715...@xxx.xxx.xxx.xxx;tag=B6534850-EC6 Date: Fri, 06 Jan 2012 04:51:39 GMT Call-ID: 7e54da423b0e6e457475ab17694e5...@yyy.yy.yy.yy Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO,
