[Asterisk-Users] Re: SIP RTP Negotiation

Tue Jan 24 21:09:22 MST 2006

Hi Asterisk-Users,

Please help as it is out of my league and understand as why the call would
be silent or partly silent (calling party can't hear the called party, but
called party can hear).  Thanks in advance.

Regards,
Kengie

On 1/19/06, Kenige Ho <kengiepanda at gmail.com> wrote:
>
> Dear All,
>
> I am having some problems with connecting with a UA.  Sometimes there is
> not sound in the call made, sometimes the caller would near no sound, while
> the callee can hear the caller.  I have attached the rtp debug and sip debug
> for you comments.  Please help me.  Thank you all.
>
> Asterisk Version is 1.2.1
> Asterisk RTP Range is 10000 to 20000
> UA Listen RTP Port is 15000
>
> Below is the the SIP Logs
>
> TestServer*CLI>
> <-- SIP read from 66.193.155.2:46478:
> REGISTER sip:XXX.XXX.XX.XXX:5060 SIP/2.0
> Via: SIP/2.0/UDP 172.28.174.25:5060
> From: XXXXXX <sip:XXXXXX at XXX.XXX.XX.XXX>
> To: XXXXXX <sip:XXXXXX at XXX.XXX.XX.XXX>
> Call-ID: 74494a-1654e-43ce24ec at XXX.XXX.XX.XXX
> CSeq: 1 REGISTER
> Contact: "XXXXXX" <sip:XXXXXX at 172.28.174.25:5060>
> User-Agent: XXXXX UserAgent/1.0
> Expires: 5000
> Max-Forwards: 70
> Content-Length: 0
>
>
> --- (11 headers 0 lines)---
> Using latest REGISTER request as basis request
> Sending to 172.28.174.25 : 5060 (non-NAT)
> Transmitting (NAT) to 66.193.155.2:46478:
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 172.28.174.25:5060;received=66.193.155.2
> From: XXXXXX < sip:XXXXXX at XXX.XXX.XX.XXX>
> To: XXXXXX <sip:XXXXXX at XXX.XXX.XX.XXX>
> Call-ID: 74494a-1654e-43ce24ec at XXX.XXX.XX.XXX
> CSeq: 1 REGISTER
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Max-Forwards: 70
> Contact: <sip:XXXXXX at XXX.XXX.XX.XXX >
> Content-Length: 0
>
>
> ---
> Transmitting (NAT) to 66.193.155.2:46478:
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/UDP 172.28.174.25:5060;received= 66.193.155.2
> From: XXXXXX <sip:XXXXXX at XXX.XXX.XX.XXX>
> To: XXXXXX <sip:XXXXXX at XXX.XXX.XX.XXX>;tag=as504de7b8
> Call-ID: 74494a-1654e-43ce24ec at XXX.XXX.XX.XXX
> CSeq: 1 REGISTER
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Max-Forwards: 70
> Contact: <sip:XXXXXX at XXX.XXX.XX.XXX>
> WWW-Authenticate: Digest realm="asterisk", nonce="31e7aa76"
> Content-Length: 0
>
>
> ---
> Scheduling destruction of call '74494a-1654e-43ce24ec at XXX.XXX.XX.XXX' in
> 15000 ms
> TestServer*CLI>
> <-- SIP read from 66.193.155.2:46478:
> REGISTER sip:XXX.XXX.XX.XXX:5060 SIP/2.0
> Via: SIP/2.0/UDP 172.28.174.25:5060
> From: XXXXXX <sip:XXXXXX at XXX.XXX.XX.XXX >
> To: XXXXXX <sip:XXXXXX at XXX.XXX.XX.XXX>
> Call-ID: 74494a-1654e-43ce24ec at XXX.XXX.XX.XXX
> CSeq: 2 REGISTER
> Contact: "XXXXXX" <sip:XXXXXX at 172.28.174.25:5060>
> Authorization: Digest username="XXXXXX", realm="asterisk",
> nonce="31e7aa76", uri="sip:XXX.XXX.XX.XXX",
> response="3e54ea5e3a3b6df1e5db7b4a3182e18f", algorithm=MD5
> User-Agent: XXXXX UserAgent/1.0
> Expires: 5000
> Max-Forwards: 70
> Content-Length: 0
>
>
> --- (12 headers 0 lines)---
> Using latest REGISTER request as basis request
> Sending to 172.28.174.25 : 5060 (NAT)
> Transmitting (NAT) to 66.193.155.2:46478:
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 172.28.174.25:5060;received=66.193.155.2
> From: XXXXXX < sip:XXXXXX at XXX.XXX.XX.XXX>
> To: XXXXXX <sip:XXXXXX at XXX.XXX.XX.XXX>
> Call-ID: 74494a-1654e-43ce24ec at XXX.XXX.XX.XXX
> CSeq: 2 REGISTER
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Max-Forwards: 70
> Contact: <sip:XXXXXX at XXX.XXX.XX.XXX >
> Content-Length: 0
>
>
> ---
> Transmitting (NAT) to 66.193.155.2:46478:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 172.28.174.25:5060;received= 66.193.155.2
> From: XXXXXX <sip:XXXXXX at XXX.XXX.XX.XXX>
> To: XXXXXX <sip:XXXXXX at XXX.XXX.XX.XXX>;tag=as504de7b8
> Call-ID: 74494a-1654e-43ce24ec at XXX.XXX.XX.XXX
> CSeq: 2 REGISTER
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Max-Forwards: 70
> Expires: 3600
> Contact: <sip:XXXXXX at 172.28.174.25:5060>;expires=3600
> Date: Wed, 18 Jan 2006 11:20:15 GMT
> Content-Length: 0
>
>
> ---
> Scheduling destruction of call '74494a-1654e-43ce24ec at XXX.XXX.XX.XXX' in
> 15000 ms
> Destroying call '4CEFE29BD3864FFFA788C2D43F4E6FBD at wawan'
> TestServer*CLI>
> <-- SIP read from 202.83.167.103:30928:
>
>
> --- (0 headers 0 lines) Nat keepalive ---
> TestServer*CLI>
> <-- SIP read from 218.111.26.5:5060:
>
>
> --- (0 headers 0 lines) Nat keepalive ---
> Destroying call 'IvDHn41FxnNwCePq at 192.168.1.35'
> TestServer*CLI>
> <-- SIP read from 66.193.155.2:46478:
> INVITE sip:123456788 at XXX.XXX.XX.XXX:5060 SIP/2.0
> Via: SIP/2.0/UDP 172.28.174.25:5060
> From: XXXXXX <sip:XXXXXX at XXX.XXX.XX.XXX >;tag=174cf52
> To: <sip:123456788 at XXX.XXX.XX.XXX>
> Call-ID: 7469ea-26568-43ce24f4 at 172.28.174.25
> CSeq: 3 INVITE
> Contact: <sip:XXXXXX at 172.28.174.25:5060>
> Subject: no subject
> User-Agent: XXXXX UserAgent/1.0
> Max-Forwards: 70
> Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
> Content-Type: application/SDP
> Accept: application/SDP, text/plain
> Accept-Encoding: identity
> Content-Length:   287
>
> v=0
> o=XXXXXX 11375833482 11375833482 IN IP4 172.28.174.25
> s=VaxSoft Inc.
> c=IN IP4 172.28.174.25
> t=0 0
> m=audio 15000 RTP/AVP 3 98 8 0 101
> a=rtpmap:3 GSM/8000
> a=rtpmap:98 iLBC/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
>
> --- (15 headers 12 lines)---
> Using INVITE request as basis request -
> 7469ea-26568-43ce24f4 at 172.28.174.25
> Sending to 172.28.174.25 : 5060 (non-NAT)
> Reliably Transmitting (NAT) to 66.193.155.2:46478:
> SIP/2.0 407 Proxy Authentication Required
> Via: SIP/2.0/UDP 172.28.174.25:5060 ;received=66.193.155.2
> From: XXXXXX <sip:XXXXXX at XXX.XXX.XX.XXX>;tag=174cf52
> To: < sip:123456788 at XXX.XXX.XX.XXX>;tag=as733e9adc
> Call-ID: 7469ea-26568-43ce24f4 at 172.28.174.25
> CSeq: 3 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Max-Forwards: 70
> Contact: <sip:123456788 at XXX.XXX.XX.XXX>
> Proxy-Authenticate: Digest realm="asterisk", nonce="452293c4"
> Content-Length: 0
>
>
> ---
> Scheduling destruction of call '7469ea-26568-43ce24f4 at 172.28.174.25' in
> 15000 ms
> Found user 'XXXXXX'
> TestServer*CLI>
> <-- SIP read from 66.193.155.2:46478:
> ACK sip:123456788 at XXX.XXX.XX.XXX:5060 SIP/2.0
> Via: SIP/2.0/UDP 172.28.174.25:5060
> From: XXXXXX < sip:XXXXXX at XXX.XXX.XX.XXX>;tag=174cf52
> To: <sip:123456788 at XXX.XXX.XX.XXX>;tag=as733e9adc
> Call-ID: 7469ea-26568-43ce24f4 at 172.28.174.25
> CSeq: 3 ACK
> User-Agent: XXXXX UserAgent/1.0
> Content-Length: 0
>
>
> --- (8 headers 0 lines)---
> TestServer*CLI>
> <-- SIP read from 66.193.155.2:46478:
> INVITE sip:123456788 at XXX.XXX.XX.XXX:5060 SIP/2.0
> Via: SIP/2.0/UDP 172.28.174.25:5060
> From: XXXXXX <sip:XXXXXX at XXX.XXX.XX.XXX>;tag=174cf52
> To: <sip:123456788 at XXX.XXX.XX.XXX >
> Call-ID: 7469ea-26568-43ce24f4 at 172.28.174.25
> CSeq: 4 INVITE
> Contact: <sip:XXXXXX at 172.28.174.25:5060>
> Proxy-Authorization: Digest username="XXXXXX", realm="asterisk",
> nonce="452293c4", uri=" sip:123456788 at XXX.XXX.XX.XXX",
> response="fd3b9b4d26c57a36ca4b1e96cca350ff", algorithm=MD5
> Subject: no subject
> User-Agent: XXXXX UserAgent/1.0
> Max-Forwards: 70
> Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
> Content-Type: application/SDP
> Accept: application/SDP, text/plain
> Accept-Encoding: identity
> Content-Length:   287
>
> v=0
> o=XXXXXX 11375833482 11375833482 IN IP4 172.28.174.25
> s=VaxSoft Inc.
> c=IN IP4 172.28.174.25
> t=0 0
> m=audio 15000 RTP/AVP 3 98 8 0 101
> a=rtpmap:3 GSM/8000
> a=rtpmap:98 iLBC/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
>
> --- (16 headers 12 lines)---
> Using INVITE request as basis request -
> 7469ea-26568-43ce24f4 at 172.28.174.25
> Sending to 172.28.174.25 : 5060 (NAT)
> Found user 'XXXXXX'
> Found RTP audio format 3
> Found RTP audio format 98
> Found RTP audio format 8
> Found RTP audio format 0
> Found RTP audio format 101
> Peer audio RTP is at port 172.28.174.25:15000
> Peer video RTP is at port 172.28.174.25:65535
> Found description format GSM
> Found description format iLBC
> Found description format PCMA
> Found description format PCMU
> Found description format telephone-event
> Capabilities: us - 0x40e (gsm|ulaw|alaw|ilbc), peer - audio=0x40e
> (gsm|ulaw|alaw|ilbc)/video=0x0 (nothing), combined - 0x40e
> (gsm|ulaw|alaw|ilbc)
> Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
> (telephone-event), combined - 0x1 (telephone-event)
> Looking for 123456788 in prepaid (domain XXX.XXX.XX.XXX)
> list_route: hop: <sip:XXXXXX at 172.28.174.25:5060>
> Transmitting (NAT) to 66.193.155.2:46478:
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 172.28.174.25:5060;received=66.193.155.2
> From: XXXXXX <sip:XXXXXX at XXX.XXX.XX.XXX >;tag=174cf52
> To: <sip:123456788 at XXX.XXX.XX.XXX>
> Call-ID: 7469ea-26568-43ce24f4 at 172.28.174.25
> CSeq: 4 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Max-Forwards: 70
> Contact: <sip:123456788 at XXX.XXX.XX.XXX >
> Content-Length: 0
>
>
> ---
> We're at XXX.XXX.XX.XXX port 17532
> Video is at XXX.XXX.XX.XXX port 14338
> Adding codec 0x400 (ilbc) to SDP
> Adding codec 0x2 (gsm) to SDP
> Adding codec 0x4 (ulaw) to SDP
> Adding codec 0x8 (alaw) to SDP
> Adding non-codec 0x1 (telephone-event) to SDP
> Reliably Transmitting (NAT) to 66.193.155.2:46478:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 172.28.174.25:5060 ;received=66.193.155.2
> From: XXXXXX <sip:XXXXXX at XXX.XXX.XX.XXX>;tag=174cf52
> To: < sip:123456788 at XXX.XXX.XX.XXX>;tag=as2894778b
> Call-ID: 7469ea-26568-43ce24f4 at 172.28.174.25
> CSeq: 4 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Max-Forwards: 70
> Contact: <sip:123456788 at XXX.XXX.XX.XXX>
> Content-Type: application/sdp
> Content-Length: 291
> TestServer*CLI>
> v=0
> o=root 4095 4095 IN IP4 XXX.XXX.XX.XXX
> s=session
> c=IN IP4 XXX.XXX.XX.XXX
> t=0 0
> m=audio 17532 RTP/AVP 98 3 0 8 101
> a=rtpmap:98 iLBC/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
>
> TestServer*CLI>
> <-- SIP read from 66.193.155.2:46478:
> ACK sip:123456788 at XXX.XXX.XX.XXX:5060 SIP/2.0
> Via: SIP/2.0/UDP 172.28.174.25:5060
> From: XXXXXX <sip:XXXXXX at XXX.XXX.XX.XXX>;tag=174cf52
> To: <sip:123456788 at XXX.XXX.XX.XXX>;tag=as2894778b
> Call-ID: 7469ea-26568-43ce24f4 at 172.28.174.25
> CSeq: 4 ACK
> User-Agent: XXXXX UserAgent/1.0
> Content-Length: 0
>
>
> Below is the RTP Logs:
>
> Sent RTP packet to 66.193.155.2:15000 (type 3, seq 37407, ts 878720, len
> 33)
> Got RTP packet from 66.193.155.2:15000 (type 3, seq 5884, ts 897565, len
> 33)
> Sent RTP packet to 66.193.155.2:15000 (type 3, seq 37408, ts 878880, len
> 33)
> Got RTP packet from 66.193.155.2:15000 (type 3, seq 5885, ts 897725, len
> 33)
> Sent RTP packet to 66.193.155.2:15000 (type 3, seq 37409, ts 879040, len
> 33)
>
>
> Regards,
> Kengie
>
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