It seems that the current head has actually started producing this "rtpmap: 96 /0", when my client devices do not include it in the sdp. This causes problems when I am trying to upgrade one switch at a time in my production environment. I assume (will test here shortly) that if all switches are on the newer version, this wouldn't be an issue as the indicated patch should handle it. I would really like to be able to upgrade my boxes individually, but if an upgraded box add's this to the SDP, my non-upgraded boxes will kill the call. The following is an example SDP trace from my latest head (10.01.2013) showing the SDP from the client device is normal, and then what is sent to the other freeswitch (my currently non upgraded one) includes this problem causing rtpmap. I should note that this seems to only be the case when I activate zrtp on the clients, which causes my dial plan to set proxy_media=true. I haven't opened a bug in jira, I wanted to confirm if this rtpmap is in fact a bug. If there is a good reason it is there now and isn't in my older freeswitch versions, then I may have to attempt to upgrade all at one time. recv 1072 bytes from tls/[1.1.1.1]:50272 at 13:42:16.509540: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/TLS 2.2.2.2;branch=z9hG4bK5N858D7BX46je Contact: <sip:15715550001@192.168.10.136:60068;transport=tls> From: "Test 1" <sip:17035550001@securevoice>;tag=jSmQZ67etF8pa Call-ID: 6fe45089-a6d4-1231-2aac-005056ac002a CSeq: 50067799 INVITE To: <sip:15715550001@192.168.10.136:60068 ;rinstance=A5E7AF45;transport=tls>;tag=8508A76BFA383C2109766C45F68BCEFB Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY, MESSAGE Supported: replaces, path Content-Type: application/sdp Content-Length: 534 v=0 o=- 98627 19401 IN IP4 192.168.10.136 s=menduco c=IN IP4 192.168.10.136 t=0 0 m=audio 58620 RTP/SAVP 9 101 a=rtpmap:101 telephone-event/8000 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:Df3mAFtpRclAzm98vfNWAoWqi6WaA3hfWs0e7CSi a=ptime:30 a=fmtp:101 0-15 a=zrtp-hash:1.10 99B821F2DBB96DFDC234440977DC3956441DEC6CDAE690B1B06C740F6EBC29B0 a=sendrecv m=audio 0 RTP/AVP 9 0 101 a=rtpmap:101 TELEPHONE-EVENT/8000 a=ptime:30 a=fmtp:101 0-15 a=zrtp-hash:1.10 18489BD8B5526F4C92AB6EB2A04022C25E8CB5D7E58D4E633FC4545783447F8C ------------------------------------------------------------------------ send 472 bytes to tls/[1.1.1.1]:50272 at 13:42:16.510594: ------------------------------------------------------------------------ ACK sip:15715550001@192.168.10.136:60068;transport=tls SIP/2.0 Via: SIP/2.0/TLS 2.2.2.2;branch=z9hG4bK6y1ya9QFtDX5S Max-Forwards: 70 From: "Test 1" <sip:17035550001@securevoice>;tag=jSmQZ67etF8pa To: <sip:15715550001@192.168.10.136:60068 ;rinstance=A5E7AF45;transport=tls>;tag=8508A76BFA383C2109766C45F68BCEFB Call-ID: 6fe45089-a6d4-1231-2aac-005056ac002a CSeq: 50067799 ACK Contact: <sip:mod_sofia@2.2.2.2:5061;transport=tls> Content-Length: 0 ------------------------------------------------------------------------ send 1516 bytes to tls/[10.110.11.31]:47793 at 13:42:16.516187: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/TLS 10.110.11.31:5071 ;branch=z9hG4bKD97QXp3XZgX4a;rport=47793 From: "Test 1" <sip:17035550001@10.110.11.31>;tag=DU0S42e40FNpg To: <sip:15715550001@dcprotopop1.private.sec :5071;transport=tls>;tag=tcXjHQ6gme91S Call-ID: 6fd5bd89-a6d4-1231-8291-005056ac002c CSeq: 50067799 INVITE Contact: <sip:15715550001@10.110.11.32:5071;transport=tls> User-Agent: FreeSWITCH-mod_sofia/1.5.6b+git~20131001T160636Z~6d2280df08 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 408 X-FS-Display-Name: Outbound Call X-FS-Display-Number: sip:15715550001@dcprotopop1.private.sec X-FS-Support: update_display,send_info Remote-Party-ID: "Outbound Call" <sip:15715550001@dcprotopop1.private.sec
;party=calling;privacy=off;screen=no
v=0 o=FreeSWITCH 1167543025 1167543026 IN IP4 10.110.11.32 s=FreeSWITCH c=IN IP4 10.110.11.32 t=0 0 m=audio 27868 RTP/SAVP 9 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:Df3mAFtpRclAzm98vfNWAoWqi6WaA3hfWs0e7CSi a=ptime:30 a=zrtp-hash:1.10 99B821F2DBB96DFDC234440977DC3956441DEC6CDAE690B1B06C740F6EBC29B0 m=audio 0 RTP/AVP 9 0 96 a=rtpmap:96 /0
I think Anthony put a patch on HEAD for this - the RFC states that this is acceptable (the a= line should be ignored when the m= line has no port), but there's a regex in recent versions of freeswitch that gets upset when it can't match anything in front of the /0 On Fri, Oct 4, 2013 at 2:54 AM, sangdrax8 <sangdrax8@gmail.com> wrote:
It seems that the current head has actually started producing this "rtpmap: 96 /0", when my client devices do not include it in the sdp. This causes problems when I am trying to upgrade one switch at a time in my production environment. I assume (will test here shortly) that if all switches are on the newer version, this wouldn't be an issue as the indicated patch should handle it.
I would really like to be able to upgrade my boxes individually, but if an upgraded box add's this to the SDP, my non-upgraded boxes will kill the call.
The following is an example SDP trace from my latest head (10.01.2013) showing the SDP from the client device is normal, and then what is sent to the other freeswitch (my currently non upgraded one) includes this problem causing rtpmap. I should note that this seems to only be the case when I activate zrtp on the clients, which causes my dial plan to set proxy_media=true.
I haven't opened a bug in jira, I wanted to confirm if this rtpmap is in fact a bug. If there is a good reason it is there now and isn't in my older freeswitch versions, then I may have to attempt to upgrade all at one time.
recv 1072 bytes from tls/[1.1.1.1]:50272 at 13:42:16.509540: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/TLS 2.2.2.2;branch=z9hG4bK5N858D7BX46je Contact: <sip:15715550001@192.168.10.136:60068;transport=tls> From: "Test 1" <sip:17035550001@securevoice>;tag=jSmQZ67etF8pa Call-ID: 6fe45089-a6d4-1231-2aac-005056ac002a CSeq: 50067799 INVITE To: <sip:15715550001@192.168.10.136:60068 ;rinstance=A5E7AF45;transport=tls>;tag=8508A76BFA383C2109766C45F68BCEFB Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY, MESSAGE Supported: replaces, path Content-Type: application/sdp Content-Length: 534
v=0 o=- 98627 19401 IN IP4 192.168.10.136 s=menduco c=IN IP4 192.168.10.136 t=0 0 m=audio 58620 RTP/SAVP 9 101 a=rtpmap:101 telephone-event/8000 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:Df3mAFtpRclAzm98vfNWAoWqi6WaA3hfWs0e7CSi a=ptime:30 a=fmtp:101 0-15 a=zrtp-hash:1.10 99B821F2DBB96DFDC234440977DC3956441DEC6CDAE690B1B06C740F6EBC29B0 a=sendrecv m=audio 0 RTP/AVP 9 0 101 a=rtpmap:101 TELEPHONE-EVENT/8000 a=ptime:30 a=fmtp:101 0-15 a=zrtp-hash:1.10 18489BD8B5526F4C92AB6EB2A04022C25E8CB5D7E58D4E633FC4545783447F8C ------------------------------------------------------------------------ send 472 bytes to tls/[1.1.1.1]:50272 at 13:42:16.510594: ------------------------------------------------------------------------ ACK sip:15715550001@192.168.10.136:60068;transport=tls SIP/2.0 Via: SIP/2.0/TLS 2.2.2.2;branch=z9hG4bK6y1ya9QFtDX5S Max-Forwards: 70 From: "Test 1" <sip:17035550001@securevoice>;tag=jSmQZ67etF8pa To: <sip:15715550001@192.168.10.136:60068 ;rinstance=A5E7AF45;transport=tls>;tag=8508A76BFA383C2109766C45F68BCEFB Call-ID: 6fe45089-a6d4-1231-2aac-005056ac002a CSeq: 50067799 ACK Contact: <sip:mod_sofia@2.2.2.2:5061;transport=tls> Content-Length: 0 ------------------------------------------------------------------------ send 1516 bytes to tls/[10.110.11.31]:47793 at 13:42:16.516187: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/TLS 10.110.11.31:5071 ;branch=z9hG4bKD97QXp3XZgX4a;rport=47793 From: "Test 1" <sip:17035550001@10.110.11.31>;tag=DU0S42e40FNpg To: <sip:15715550001 @dcprotopop1.private.sec:5071;transport=tls>;tag=tcXjHQ6gme91S Call-ID: 6fd5bd89-a6d4-1231-8291-005056ac002c CSeq: 50067799 INVITE Contact: <sip:15715550001@10.110.11.32:5071;transport=tls> User-Agent: FreeSWITCH-mod_sofia/1.5.6b+git~20131001T160636Z~6d2280df08 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 408 X-FS-Display-Name: Outbound Call X-FS-Display-Number: sip:15715550001@dcprotopop1.private.sec X-FS-Support: update_display,send_info Remote-Party-ID: "Outbound Call" <sip:15715550001 @dcprotopop1.private.sec>;party=calling;privacy=off;screen=no
v=0 o=FreeSWITCH 1167543025 1167543026 IN IP4 10.110.11.32 s=FreeSWITCH c=IN IP4 10.110.11.32 t=0 0 m=audio 27868 RTP/SAVP 9 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:Df3mAFtpRclAzm98vfNWAoWqi6WaA3hfWs0e7CSi a=ptime:30 a=zrtp-hash:1.10 99B821F2DBB96DFDC234440977DC3956441DEC6CDAE690B1B06C740F6EBC29B0 m=audio 0 RTP/AVP 9 0 96 a=rtpmap:96 /0
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Yes head does handle this with out hanging up, but it also inserts it now when the client didn't. So I was wondering why it was adding this which breaks comparability with all older versions of FreeSWITCH which don't insert this. On Oct 3, 2013 6:03 PM, "Sam Russell" <sam.h.russell@gmail.com> wrote:
I think Anthony put a patch on HEAD for this - the RFC states that this is acceptable (the a= line should be ignored when the m= line has no port), but there's a regex in recent versions of freeswitch that gets upset when it can't match anything in front of the /0
On Fri, Oct 4, 2013 at 2:54 AM, sangdrax8 <sangdrax8@gmail.com> wrote:
It seems that the current head has actually started producing this "rtpmap: 96 /0", when my client devices do not include it in the sdp. This causes problems when I am trying to upgrade one switch at a time in my production environment. I assume (will test here shortly) that if all switches are on the newer version, this wouldn't be an issue as the indicated patch should handle it.
I would really like to be able to upgrade my boxes individually, but if an upgraded box add's this to the SDP, my non-upgraded boxes will kill the call.
The following is an example SDP trace from my latest head (10.01.2013) showing the SDP from the client device is normal, and then what is sent to the other freeswitch (my currently non upgraded one) includes this problem causing rtpmap. I should note that this seems to only be the case when I activate zrtp on the clients, which causes my dial plan to set proxy_media=true.
I haven't opened a bug in jira, I wanted to confirm if this rtpmap is in fact a bug. If there is a good reason it is there now and isn't in my older freeswitch versions, then I may have to attempt to upgrade all at one time.
recv 1072 bytes from tls/[1.1.1.1]:50272 at 13:42:16.509540:
------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/TLS 2.2.2.2;branch=z9hG4bK5N858D7BX46je Contact: <sip:15715550001@192.168.10.136:60068;transport=tls> From: "Test 1" <sip:17035550001@securevoice>;tag=jSmQZ67etF8pa Call-ID: 6fe45089-a6d4-1231-2aac-005056ac002a CSeq: 50067799 INVITE To: <sip:15715550001@192.168.10.136:60068 ;rinstance=A5E7AF45;transport=tls>;tag=8508A76BFA383C2109766C45F68BCEFB Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY, MESSAGE Supported: replaces, path Content-Type: application/sdp Content-Length: 534
v=0 o=- 98627 19401 IN IP4 192.168.10.136 s=menduco c=IN IP4 192.168.10.136 t=0 0 m=audio 58620 RTP/SAVP 9 101 a=rtpmap:101 telephone-event/8000 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:Df3mAFtpRclAzm98vfNWAoWqi6WaA3hfWs0e7CSi a=ptime:30 a=fmtp:101 0-15 a=zrtp-hash:1.10 99B821F2DBB96DFDC234440977DC3956441DEC6CDAE690B1B06C740F6EBC29B0 a=sendrecv m=audio 0 RTP/AVP 9 0 101 a=rtpmap:101 TELEPHONE-EVENT/8000 a=ptime:30 a=fmtp:101 0-15 a=zrtp-hash:1.10 18489BD8B5526F4C92AB6EB2A04022C25E8CB5D7E58D4E633FC4545783447F8C
------------------------------------------------------------------------ send 472 bytes to tls/[1.1.1.1]:50272 at 13:42:16.510594:
------------------------------------------------------------------------ ACK sip:15715550001@192.168.10.136:60068;transport=tls SIP/2.0 Via: SIP/2.0/TLS 2.2.2.2;branch=z9hG4bK6y1ya9QFtDX5S Max-Forwards: 70 From: "Test 1" <sip:17035550001@securevoice>;tag=jSmQZ67etF8pa To: <sip:15715550001@192.168.10.136:60068 ;rinstance=A5E7AF45;transport=tls>;tag=8508A76BFA383C2109766C45F68BCEFB Call-ID: 6fe45089-a6d4-1231-2aac-005056ac002a CSeq: 50067799 ACK Contact: <sip:mod_sofia@2.2.2.2:5061;transport=tls> Content-Length: 0 ------------------------------------------------------------------------ send 1516 bytes to tls/[10.110.11.31]:47793 at 13:42:16.516187:
------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/TLS 10.110.11.31:5071 ;branch=z9hG4bKD97QXp3XZgX4a;rport=47793 From: "Test 1" <sip:17035550001@10.110.11.31>;tag=DU0S42e40FNpg To: <sip:15715550001 @dcprotopop1.private.sec:5071;transport=tls>;tag=tcXjHQ6gme91S Call-ID: 6fd5bd89-a6d4-1231-8291-005056ac002c CSeq: 50067799 INVITE Contact: <sip:15715550001@10.110.11.32:5071;transport=tls> User-Agent: FreeSWITCH-mod_sofia/1.5.6b+git~20131001T160636Z~6d2280df08 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 408 X-FS-Display-Name: Outbound Call X-FS-Display-Number: sip:15715550001@dcprotopop1.private.sec X-FS-Support: update_display,send_info Remote-Party-ID: "Outbound Call" <sip:15715550001 @dcprotopop1.private.sec>;party=calling;privacy=off;screen=no
v=0 o=FreeSWITCH 1167543025 1167543026 IN IP4 10.110.11.32 s=FreeSWITCH c=IN IP4 10.110.11.32 t=0 0 m=audio 27868 RTP/SAVP 9 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:Df3mAFtpRclAzm98vfNWAoWqi6WaA3hfWs0e7CSi a=ptime:30 a=zrtp-hash:1.10 99B821F2DBB96DFDC234440977DC3956441DEC6CDAE690B1B06C740F6EBC29B0 m=audio 0 RTP/AVP 9 0 96 a=rtpmap:96 /0
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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Do you have any logs from this? I have not seen this before and I am not sure what would be causing that. Full console log with sip traces. POST on JIRA On Thu, Oct 3, 2013 at 6:00 PM, sangdrax8 <sangdrax8@gmail.com> wrote:
Yes head does handle this with out hanging up, but it also inserts it now when the client didn't. So I was wondering why it was adding this which breaks comparability with all older versions of FreeSWITCH which don't insert this. On Oct 3, 2013 6:03 PM, "Sam Russell" <sam.h.russell@gmail.com> wrote:
I think Anthony put a patch on HEAD for this - the RFC states that this is acceptable (the a= line should be ignored when the m= line has no port), but there's a regex in recent versions of freeswitch that gets upset when it can't match anything in front of the /0
On Fri, Oct 4, 2013 at 2:54 AM, sangdrax8 <sangdrax8@gmail.com> wrote:
It seems that the current head has actually started producing this "rtpmap: 96 /0", when my client devices do not include it in the sdp. This causes problems when I am trying to upgrade one switch at a time in my production environment. I assume (will test here shortly) that if all switches are on the newer version, this wouldn't be an issue as the indicated patch should handle it.
I would really like to be able to upgrade my boxes individually, but if an upgraded box add's this to the SDP, my non-upgraded boxes will kill the call.
The following is an example SDP trace from my latest head (10.01.2013) showing the SDP from the client device is normal, and then what is sent to the other freeswitch (my currently non upgraded one) includes this problem causing rtpmap. I should note that this seems to only be the case when I activate zrtp on the clients, which causes my dial plan to set proxy_media=true.
I haven't opened a bug in jira, I wanted to confirm if this rtpmap is in fact a bug. If there is a good reason it is there now and isn't in my older freeswitch versions, then I may have to attempt to upgrade all at one time.
recv 1072 bytes from tls/[1.1.1.1]:50272 at 13:42:16.509540:
------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/TLS 2.2.2.2;branch=z9hG4bK5N858D7BX46je Contact: <sip:15715550001@192.168.10.136:60068;transport=tls> From: "Test 1" <sip:17035550001@securevoice>;tag=jSmQZ67etF8pa Call-ID: 6fe45089-a6d4-1231-2aac-005056ac002a CSeq: 50067799 INVITE To: <sip:15715550001@192.168.10.136:60068 ;rinstance=A5E7AF45;transport=tls>;tag=8508A76BFA383C2109766C45F68BCEFB Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY, MESSAGE Supported: replaces, path Content-Type: application/sdp Content-Length: 534
v=0 o=- 98627 19401 IN IP4 192.168.10.136 s=menduco c=IN IP4 192.168.10.136 t=0 0 m=audio 58620 RTP/SAVP 9 101 a=rtpmap:101 telephone-event/8000 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:Df3mAFtpRclAzm98vfNWAoWqi6WaA3hfWs0e7CSi a=ptime:30 a=fmtp:101 0-15 a=zrtp-hash:1.10 99B821F2DBB96DFDC234440977DC3956441DEC6CDAE690B1B06C740F6EBC29B0 a=sendrecv m=audio 0 RTP/AVP 9 0 101 a=rtpmap:101 TELEPHONE-EVENT/8000 a=ptime:30 a=fmtp:101 0-15 a=zrtp-hash:1.10 18489BD8B5526F4C92AB6EB2A04022C25E8CB5D7E58D4E633FC4545783447F8C
------------------------------------------------------------------------ send 472 bytes to tls/[1.1.1.1]:50272 at 13:42:16.510594:
------------------------------------------------------------------------ ACK sip:15715550001@192.168.10.136:60068;transport=tls SIP/2.0 Via: SIP/2.0/TLS 2.2.2.2;branch=z9hG4bK6y1ya9QFtDX5S Max-Forwards: 70 From: "Test 1" <sip:17035550001@securevoice>;tag=jSmQZ67etF8pa To: <sip:15715550001@192.168.10.136:60068 ;rinstance=A5E7AF45;transport=tls>;tag=8508A76BFA383C2109766C45F68BCEFB Call-ID: 6fe45089-a6d4-1231-2aac-005056ac002a CSeq: 50067799 ACK Contact: <sip:mod_sofia@2.2.2.2:5061;transport=tls> Content-Length: 0 ------------------------------------------------------------------------ send 1516 bytes to tls/[10.110.11.31]:47793 at 13:42:16.516187:
------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/TLS 10.110.11.31:5071 ;branch=z9hG4bKD97QXp3XZgX4a;rport=47793 From: "Test 1" <sip:17035550001@10.110.11.31>;tag=DU0S42e40FNpg To: <sip:15715550001 @dcprotopop1.private.sec:5071;transport=tls>;tag=tcXjHQ6gme91S Call-ID: 6fd5bd89-a6d4-1231-8291-005056ac002c CSeq: 50067799 INVITE Contact: <sip:15715550001@10.110.11.32:5071;transport=tls> User-Agent: FreeSWITCH-mod_sofia/1.5.6b+git~20131001T160636Z~6d2280df08 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 408 X-FS-Display-Name: Outbound Call X-FS-Display-Number: sip:15715550001@dcprotopop1.private.sec X-FS-Support: update_display,send_info Remote-Party-ID: "Outbound Call" <sip:15715550001 @dcprotopop1.private.sec>;party=calling;privacy=off;screen=no
v=0 o=FreeSWITCH 1167543025 1167543026 IN IP4 10.110.11.32 s=FreeSWITCH c=IN IP4 10.110.11.32 t=0 0 m=audio 27868 RTP/SAVP 9 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:Df3mAFtpRclAzm98vfNWAoWqi6WaA3hfWs0e7CSi a=ptime:30 a=zrtp-hash:1.10 99B821F2DBB96DFDC234440977DC3956441DEC6CDAE690B1B06C740F6EBC29B0 m=audio 0 RTP/AVP 9 0 96 a=rtpmap:96 /0
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-dev mailing list FreeSWITCH-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-dev mailing list FreeSWITCH-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-dev mailing list FreeSWITCH-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
Sorry was away from work a few days, but I should still have the setup which these sip traces came from. I will turn on full console logging and create a Jira ticket with the information asap. Thank you! On Fri, Oct 4, 2013 at 6:49 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
Do you have any logs from this? I have not seen this before and I am not sure what would be causing that. Full console log with sip traces. POST on JIRA
On Thu, Oct 3, 2013 at 6:00 PM, sangdrax8 <sangdrax8@gmail.com> wrote:
Yes head does handle this with out hanging up, but it also inserts it now when the client didn't. So I was wondering why it was adding this which breaks comparability with all older versions of FreeSWITCH which don't insert this. On Oct 3, 2013 6:03 PM, "Sam Russell" <sam.h.russell@gmail.com> wrote:
I think Anthony put a patch on HEAD for this - the RFC states that this is acceptable (the a= line should be ignored when the m= line has no port), but there's a regex in recent versions of freeswitch that gets upset when it can't match anything in front of the /0
On Fri, Oct 4, 2013 at 2:54 AM, sangdrax8 <sangdrax8@gmail.com> wrote:
It seems that the current head has actually started producing this "rtpmap: 96 /0", when my client devices do not include it in the sdp. This causes problems when I am trying to upgrade one switch at a time in my production environment. I assume (will test here shortly) that if all switches are on the newer version, this wouldn't be an issue as the indicated patch should handle it.
I would really like to be able to upgrade my boxes individually, but if an upgraded box add's this to the SDP, my non-upgraded boxes will kill the call.
The following is an example SDP trace from my latest head (10.01.2013) showing the SDP from the client device is normal, and then what is sent to the other freeswitch (my currently non upgraded one) includes this problem causing rtpmap. I should note that this seems to only be the case when I activate zrtp on the clients, which causes my dial plan to set proxy_media=true.
I haven't opened a bug in jira, I wanted to confirm if this rtpmap is in fact a bug. If there is a good reason it is there now and isn't in my older freeswitch versions, then I may have to attempt to upgrade all at one time.
recv 1072 bytes from tls/[1.1.1.1]:50272 at 13:42:16.509540:
------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/TLS 2.2.2.2;branch=z9hG4bK5N858D7BX46je Contact: <sip:15715550001@192.168.10.136:60068;transport=tls> From: "Test 1" <sip:17035550001@securevoice>;tag=jSmQZ67etF8pa Call-ID: 6fe45089-a6d4-1231-2aac-005056ac002a CSeq: 50067799 INVITE To: <sip:15715550001@192.168.10.136:60068 ;rinstance=A5E7AF45;transport=tls>;tag=8508A76BFA383C2109766C45F68BCEFB Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY, MESSAGE Supported: replaces, path Content-Type: application/sdp Content-Length: 534
v=0 o=- 98627 19401 IN IP4 192.168.10.136 s=menduco c=IN IP4 192.168.10.136 t=0 0 m=audio 58620 RTP/SAVP 9 101 a=rtpmap:101 telephone-event/8000 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:Df3mAFtpRclAzm98vfNWAoWqi6WaA3hfWs0e7CSi a=ptime:30 a=fmtp:101 0-15 a=zrtp-hash:1.10 99B821F2DBB96DFDC234440977DC3956441DEC6CDAE690B1B06C740F6EBC29B0 a=sendrecv m=audio 0 RTP/AVP 9 0 101 a=rtpmap:101 TELEPHONE-EVENT/8000 a=ptime:30 a=fmtp:101 0-15 a=zrtp-hash:1.10 18489BD8B5526F4C92AB6EB2A04022C25E8CB5D7E58D4E633FC4545783447F8C
------------------------------------------------------------------------ send 472 bytes to tls/[1.1.1.1]:50272 at 13:42:16.510594:
------------------------------------------------------------------------ ACK sip:15715550001@192.168.10.136:60068;transport=tls SIP/2.0 Via: SIP/2.0/TLS 2.2.2.2;branch=z9hG4bK6y1ya9QFtDX5S Max-Forwards: 70 From: "Test 1" <sip:17035550001@securevoice>;tag=jSmQZ67etF8pa To: <sip:15715550001@192.168.10.136:60068 ;rinstance=A5E7AF45;transport=tls>;tag=8508A76BFA383C2109766C45F68BCEFB Call-ID: 6fe45089-a6d4-1231-2aac-005056ac002a CSeq: 50067799 ACK Contact: <sip:mod_sofia@2.2.2.2:5061;transport=tls> Content-Length: 0 ------------------------------------------------------------------------ send 1516 bytes to tls/[10.110.11.31]:47793 at 13:42:16.516187:
------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/TLS 10.110.11.31:5071 ;branch=z9hG4bKD97QXp3XZgX4a;rport=47793 From: "Test 1" <sip:17035550001@10.110.11.31>;tag=DU0S42e40FNpg To: <sip:15715550001 @dcprotopop1.private.sec:5071;transport=tls>;tag=tcXjHQ6gme91S Call-ID: 6fd5bd89-a6d4-1231-8291-005056ac002c CSeq: 50067799 INVITE Contact: <sip:15715550001@10.110.11.32:5071;transport=tls> User-Agent: FreeSWITCH-mod_sofia/1.5.6b+git~20131001T160636Z~6d2280df08 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 408 X-FS-Display-Name: Outbound Call X-FS-Display-Number: sip:15715550001@dcprotopop1.private.sec X-FS-Support: update_display,send_info Remote-Party-ID: "Outbound Call" <sip:15715550001 @dcprotopop1.private.sec>;party=calling;privacy=off;screen=no
v=0 o=FreeSWITCH 1167543025 1167543026 IN IP4 10.110.11.32 s=FreeSWITCH c=IN IP4 10.110.11.32 t=0 0 m=audio 27868 RTP/SAVP 9 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:Df3mAFtpRclAzm98vfNWAoWqi6WaA3hfWs0e7CSi a=ptime:30 a=zrtp-hash:1.10 99B821F2DBB96DFDC234440977DC3956441DEC6CDAE690B1B06C740F6EBC29B0 m=audio 0 RTP/AVP 9 0 96 a=rtpmap:96 /0
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-dev mailing list FreeSWITCH-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-dev mailing list FreeSWITCH-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-dev mailing list FreeSWITCH-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-dev mailing list FreeSWITCH-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org
participants (3)
-
Anthony Minessale
-
Sam Russell
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sangdrax8